2009.bib

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@inproceedings{Ehnes2009An-Automated-Me,
  author = {Ehnes, Jochen},
  doi = {10.1007/978-3-642-01347-8_79},
  title = {An Automated Meeting Assistant: A Tangible Mixed Reality Interface for the {AMIDA} Automatic Content Linking Device},
  booktitle = {ICEIS},
  pages = {952--962},
  year = {2009},
  keywords = {Tangible User Interface, Mixed Reality, AMI, Content Linking},
  bibsource = {DBLP, http://dblp.uni-trier.de},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/00240952.pdf},
  abstract = {We describe our approach to support ongoing meetings with an automated meeting assistant. The system based on the AMIDA Content Linking Device aims at providing relevant documents used in previous meetings for the ongoing meeting based on automatic speech recognition. Once the content linking device finds documents linked to a discussion about a similar subject in a previous meeting, it assumes they may be relevant for the current discussion as well. We believe that the way these documents are offered to the meeting participants is equally important as the way they are found. We developed a mixed reality, projection based user interface that lets the documents appear on the table tops in front of the meeting participants. They can hand them over to others or bring them onto the shared projection screen easily if they consider them relevant. Yet, irrelevant documents don't draw too much attention from the discussion. In this paper we describe the concept and implementation of this user interface and provide some preliminary results.},
  categories = {Tangible User Interface, Mixed Reality, AMI, Content Linking},
  crossref = {DBLP:conf/iceis/2009}
}
@article{murray2009,
  author = {Murray, Gabriel and Kleinbauer, Thomas and Poller, Peter and Becker, Tilman and Renals, Steve and Kilgour, Jonathan},
  doi = {10.1145/1596517.1596518},
  title = {Extrinsic Summarization Evaluation: A Decision Audit Task},
  url = {http://doi.acm.org/10.1145/1596517.1596518},
  journal = {ACM Transactions on Speech and Language Processing},
  number = {2},
  pages = {1--29},
  volume = {6},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/murray-acm09.pdf},
  abstract = {In this work we describe a large-scale extrinsic evaluation of automatic speech summarization technologies for meeting speech. The particular task is a decision audit, wherein a user must satisfy a complex information need, navigating several meetings in order to gain an understanding of how and why a given decision was made. We compare the usefulness of extractive and abstractive technologies in satisfying this information need, and assess the impact of automatic speech recognition (ASR) errors on user performance. We employ several evaluation methods for participant performance, including post-questionnaire data, human subjective and objective judgments, and a detailed analysis of participant browsing behavior. We find that while ASR errors affect user satisfaction on an information retrieval task, users can adapt their browsing behavior to complete the task satisfactorily. Results also indicate that users consider extractive summaries to be intuitive and useful tools for browsing multimodal meeting data. We discuss areas in which automatic summarization techniques can be improved in comparison with gold-standard meeting abstracts.}
}
@inproceedings{anderssoncabral09,
  author = {Andersson, J. Sebastian and Cabral, Joao P. and Badino, Leonardo and Yamagishi, Junichi and Clark, Robert A.J.},
  title = {Glottal Source and Prosodic Prominence Modelling in {HMM}-based Speech Synthesis for the {B}lizzard {C}hallenge 2009},
  booktitle = {The Blizzard Challenge 2009},
  address = {Edinburgh, U.K.},
  month = {September},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/cstr_Blizzard2009.pdf},
  abstract = {This paper describes the CSTR entry for the Blizzard Challenge 2009. The work focused on modifying two parts of the Nitech 2005 HTS speech synthesis system to improve naturalness and contextual appropriateness. The first part incorporated an implementation of the Linjencrants-Fant (LF) glottal source model. The second part focused on improving synthesis of prosodic prominence including emphasis through context dependent phonemes. Emphasis was assigned to the synthesised test sentences based on a handful of theory based rules. The two parts (LF-model and prosodic prominence) were not combined and hence evaluated separately. The results on naturalness for the LF-model showed that it is not yet perceived as natural as the Benchmark HTS system for neutral speech. The results for the prosodic prominence modelling showed that it was perceived as contextually appropriate as the Benchmark HTS system, despite a low naturalness score. The Blizzard challenge evaluation has provided valuable information on the status of our work and continued work will begin with analysing why our modifications resulted in reduced naturalness compared to the Benchmark HTS system.},
  categories = {HMM, HTS, speech synthesis, LF-model, glottal source, prosodic prominence, emphasis}
}
@phdthesis{cuayahuitl_thesis2009,
  author = {Cuayáhuitl, Heriberto},
  school = {School of Informatics, University of Edinburgh},
  title = {Hierarchical Reinforcement Learning for Spoken Dialogue Systems},
  abstract = {This thesis focuses on the problem of scalable optimization of dialogue behaviour in speech-based conversational systems using reinforcement learning. Most previous investigations in dialogue strategy learning have proposed flat reinforcement learning methods, which are more suitable for small-scale spoken dialogue systems. This research formulates the problem in terms of Semi-Markov Decision Processes (SMDPs), and proposes two hierarchical reinforcement learning methods to optimize sub-dialogues rather than full dialogues. The first method uses a hierarchy of SMDPs, where every SMDP ignores irrelevant state variables and actions in order to optimize a sub-dialogue. The second method extends the first one by constraining every SMDP in the hierarchy with prior expert knowledge. The latter method proposes a learning algorithm called 'HAM+HSMQ-Learning', which combines two existing algorithms in the literature of hierarchical reinforcement learning. Whilst the first method generates fully-learnt behaviour, the second one generates semi-learnt behaviour. In addition, this research proposes a heuristic dialogue simulation environment for automatic dialogue strategy learning. Experiments were performed on simulated and real environments based on a travel planning spoken dialogue system. Experimental results provided evidence to support the following claims: First, both methods scale well at the cost of near-optimal solutions, resulting in slightly longer dialogues than the optimal solutions. Second, dialogue strategies learnt with coherent user behaviour and conservative recognition error rates can outperform a reasonable hand-coded strategy. Third, semi-learnt dialogue behaviours are a better alternative (because of their higher overall performance) than hand-coded or fully-learnt dialogue behaviours. Last, hierarchical reinforcement learning dialogue agents are feasible and promising for the (semi) automatic design of adaptive behaviours in larger-scale spoken dialogue systems. This research makes the following contributions to spoken dialogue systems which learn their dialogue behaviour. First, the Semi-Markov Decision Process (SMDP) model was proposed to learn spoken dialogue strategies in a scalable way. Second, the concept of 'partially specified dialogue strategies' was proposed for integrating simultaneously hand-coded and learnt spoken dialogue behaviours into a single learning framework. Third, an evaluation with real users of hierarchical reinforcement learning dialogue agents was essential to validate their effectiveness in a realistic environment.},
  month = {January},
  key = {spoken dialogue systems, (semi-)automatic dialogue strategy design, hierarchical control, prior expert knowledge, Semi-Markov decision processes, hierarchical reinforcement learning},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/PhDThesis-HeribertoCuayahuitl-Final.pdf}
}
@incollection{vipperla2009a,
  author = {Vipperla, Ravi Chander and Wolters, Maria and Georgila, Kallirroi and Renals, Steve},
  publisher = {Springer},
  doi = {10.1007/978-3-642-02710-9},
  title = {Speech Input from Older Users in Smart Environments: Challenges and Perspectives},
  url = {http://www.springerlink.com/content/27r01345r1683251/?p=ad2394d646814db59cf9868b0f74b11e&pi=13},
  series = {Lecture Notes in Computer Science},
  booktitle = {Proc. HCI International: Universal Access in Human-Computer Interaction. Intelligent and Ubiquitous Interaction Environments},
  number = {5615},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/vipperla2009a.pdf},
  abstract = {Although older people are an important user group for smart environments, there has been relatively little work on adapting natural language interfaces to their requirements. In this paper, we focus on a particularly thorny problem: processing speech input from older users. Our experiments on the MATCH corpus show clearly that we need age-specific adaptation in order to recognize older users' speech reliably. Language models need to cover typical interaction patterns of older people, and acoustic models need to accommodate older voices. Further research is needed into intelligent adaptation techniques that will allow existing large, robust systems to be adapted with relatively small amounts of in-domain, age appropriate data. In addition, older users need to be supported with adequate strategies for handling speech recognition errors.}
}
@inproceedings{Ehnes2009A-Tangible-Mixed,
  editor = {Smith, Michael J. and Salvendy, Gavriel},
  author = {Ehnes, Jochen},
  publisher = {Springer},
  isbn = {978-3-642-02555-6},
  title = {A Tangible Mixed Reality Interface for the {AMI} Automated Meeting Assistant},
  series = {Lecture Notes in Computer Science},
  booktitle = {Human Interface and the Management of Information},
  pages = {485--494},
  volume = {5617},
  location = {Heidelberg},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/56170485.pdf},
  abstract = {In this paper we describe our approach to support ongoing meetings with an automated meeting assistant. We propose an alternative user interface for the AMIDA Content Linking Device. In order for the system to be less distractive and more collaborative than the original laptop screen based one, we developed a system that projects documents onto the table tops right in front of the meeting participants. This way they appear as if they were printed on paper, lying in front of the participants. We describe our setup as well as the user interface we built to handle and share these documents.},
  categories = {Mixed Reality, AMI, Content Linking, User Interface}
}
@inproceedings{zen:HTSoverview,
  author = {Zen, Heiga and Oura, Keiichiro and Nose, Takashi and Yamagishi, Junichi and Sako, Shinji and Toda, Tomoki and Masuko, Takashi and Black, Alan W. and Tokuda, Keiichi},
  title = {Recent development of the {HMM}-based speech synthesis system ({HTS})},
  booktitle = {Proc. 2009 Asia-Pacific Signal and Information Processing Association (APSIPA)},
  address = {Sapporo, Japan},
  month = {October},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/zen_APSIPA2009.pdf},
  abstract = {A statistical parametric approach to speech synthesis based on hidden Markov models (HMMs) has grown in popularity over the last few years. In this approach, spectrum, excitation, and duration of speech are simultaneously modeled by context-dependent HMMs, and speech waveforms are generate from the HMMs themselves. Since December 2002, we have publicly released an open-source software toolkit named “HMM-based speech synthesis system (HTS)” to provide a research and development toolkit for statistical parametric speech synthesis. This paper describes recent developments of HTS in detail, as well as future release plans.}
}
@article{cuayahuitl2009,
  author = {Cuayáhuitl, Heriberto and Renals, Steve and Lemon, Oliver and Shimodaira, Hiroshi},
  doi = {10.1016/j.csl.2009.07.001},
  title = {Evaluation of a hierarchical reinforcement learning spoken dialogue system},
  journal = {Computer Speech and Language},
  number = {2},
  pages = {395-429},
  volume = {24},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/cuayahuitl-csl09.pdf},
  abstract = {We describe an evaluation of spoken dialogue strategies designed using hierarchical reinforcement learning agents. The dialogue strategies were learnt in a simulated environment and tested in a laboratory setting with 32 users. These dialogues were used to evaluate three types of machine dialogue behaviour: hand-coded, fully-learnt and semi-learnt. These experiments also served to evaluate the realism of simulated dialogues using two proposed metrics contrasted with ‘Precision-Recall’. The learnt dialogue behaviours used the Semi-Markov Decision Process (SMDP) model, and we report the first evaluation of this model in a realistic conversational environment. Experimental results in the travel planning domain provide evidence to support the following claims: (a) hierarchical semi-learnt dialogue agents are a better alternative (with higher overall performance) than deterministic or fully-learnt behaviour; (b) spoken dialogue strategies learnt with highly coherent user behaviour and conservative recognition error rates (keyword error rate of 20\%) can outperform a reasonable hand-coded strategy; and (c) hierarchical reinforcement learning dialogue agents are feasible and promising for the (semi) automatic design of optimized dialogue behaviours in larger-scale systems.}
}
@incollection{sarah:hts09,
  editor = {Mullennix, John W. and Stern, Steven E.},
  author = {Creer, Sarah and Green, Phil and Cunningham, Stuart and Yamagishi, Junichi},
  publisher = {IGI Global},
  title = {Building personalised synthesised voices for individuals with dysarthria using the {HTS} toolkit},
  booktitle = {Computer Synthesized Speech Technologies: Tools for Aiding Impairment},
  note = {in press},
  edition = {1st},
  year = {2009},
  abstract = {When the speech of an individual becomes unintelligible due to a neurological disorder, a synthesized voice can replace that of the individual. To fully replace all functions of human speech communication: communication of information, maintenance of social relationships and displaying identity, the voice must be intelligible, natural-sounding and retain the vocal identity of the speaker. For speakers with dysarthria, achieving this output with minimal data recordings and deteriorating speech is difficult. An alternative to this is using Hidden Markov models (HMMs) which require much less speech data than needed for concatenative methods, to adapt a robust statistical model of speech towards the speaker characteristics captured in the data recorded by the individual. This chapter implements this technique using the HTS toolkit to build personalized synthetic voices for two individuals with dysarthria. An evaluation of the voices by the participants themselves suggests that this technique shows promise for building and reconstructing personalized voices for individuals with dysarthria once deterioration has begun.}
}
@inproceedings{Ayletetal09,
  author = {Aylett, Matthew P. and King, Simon and Yamagishi, Junichi},
  title = {Speech Synthesis Without a Phone Inventory},
  booktitle = {Interspeech},
  pages = {2087--2090},
  place = {Brighton},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/ma_interspeech09.pdf},
  abstract = {In speech synthesis the unit inventory is decided using phonological and phonetic expertise. This process is resource intensive and potentially sub-optimal. In this paper we investigate how acoustic clustering, together with lexicon constraints, can be used to build a self-organised inventory. Six English speech synthesis systems were built using two frameworks, unit selection and parametric HTS for three inventory conditions: 1) a traditional phone set, 2) a system using orthographic units, and 3) a self-organised inventory. A listening test showed a strong preference for the classic system, and for the orthographic system over the self-organised system. Results also varied by letter to sound complexity and database coverage. This suggests the self-organised approach failed to generalise pronunciation as well as introducing noise above and beyond that caused by orthographic sound mismatch.},
  categories = {speech synthesis, unit selection, parametric synthesis, phone inventory, orthographic synthesis}
}
@inproceedings{dongwang_interspeech09_spm,
  author = {Wang, Dong and King, Simon and Frankel, Joe},
  title = {Stochastic Pronunciation Modelling for Spoken Term Detection},
  booktitle = {Proc. Interspeech},
  address = {Brighton, UK},
  month = {September},
  pages = {2135--2138},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/spm.pdf},
  abstract = {A major challenge faced by a spoken term detection (STD) system is the detection of out-of-vocabulary (OOV) terms. Although a subword-based STD system is able to detect OOV terms, performance reduction is always observed compared to in-vocabulary terms. Current approaches to STD do not acknowledge the particular properties of OOV terms, such as pronunciation uncertainty. In this paper, we use a stochastic pronunciation model to deal with the uncertain pronunciations of OOV terms. By considering all possible term pronunciations, predicted by a joint-multigram model, we observe a significant performance improvement.},
  categories = {joint-multigram, pronunciation model, spoken term detection, speech recognition}
}
@inproceedings{johnson-aas:09,
  author = {Johnson, Christine and Campbell, Pauline and DePlacido, Christine and Liddell, Amy and Wolters, Maria},
  title = {Does Peripheral Hearing Loss Affect {RGDT} Thresholds in Older Adults},
  booktitle = {Proceedings of the {A}merican {A}uditory {S}ociety {C}onference},
  month = {March},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/aas09.pdf},
  abstract = {},
  categories = {speech synthesis, older users}
}
@inproceedings{wolters-is:09,
  author = {Wolters, Maria and Vipperla, Ravichander and Renals, Steve},
  title = {Age Recognition for Spoken Dialogue Systems: Do We Need It?},
  booktitle = {Proc. Interspeech},
  month = {September},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/is09.pdf},
  abstract = {When deciding whether to adapt relevant aspects of the system to the particular needs of older users, spoken dialogue systems often rely on automatic detection of chronological age. In this paper, we show that vocal ageing as measured by acoustic features is an unreliable indicator of the need for adaptation. Simple lexical features greatly improve the prediction of both relevant aspects of cognition and interactions style. Lexical features also boost age group prediction. We suggest that adaptation should be based on observed behaviour, not on chronological age, unless it is not feasible to build classifiers for relevant adaptation decisions.},
  categories = {age recognition, spoken dialogue systems}
}
@inproceedings{huang2009-icassp,
  author = {Huang, Songfang and Zhou, Bowen},
  title = {An {EM} Algorithm for {SCFG} in Formal Syntax-based Translation},
  booktitle = {Proc. IEEE International Conference on Acoustic, Speech, and Signal Processing (ICASSP'09)},
  address = {Taiwan, China},
  month = {April},
  pages = {4813--4816},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/icassp09.pdf},
  abstract = {In this paper, we investigate the use of bilingual parsing on parallel corpora to better estimate the rule parameters in a formal syntax-based machine translation system, which are normally estimated from the inaccurate heuristics. We use an Expectation-Maximization (EM) algorithm to re-estimate the parameters of synchronous context-free grammar (SCFG) rules according to the derivation knowledge from parallel corpora based on maximum likelihood principle, rather than using only the heuristic information. The proposed algorithm produces significantly better BLEU scores than a state-of-the-art formal syntax-based machine translation system on the IWSLT 2006 Chinese to English task.}
}
@inproceedings{huang2009-is,
  author = {Huang, Songfang and Renals, Steve},
  title = {A Parallel Training Algorithm for Hierarchical {P}itman-{Y}or Process Language Models},
  booktitle = {Proc. Interspeech'09},
  address = {Brighton, UK},
  month = {September},
  pages = {2695--2698},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/sh_interspeech09.pdf},
  abstract = {The Hierarchical Pitman Yor Process Language Model (HPYLM) is a Bayesian language model based on a non-parametric prior, the Pitman-Yor Process. It has been demonstrated, both theoretically and practically, that the HPYLM can provide better smoothing for language modeling, compared with state-of-the-art approaches such as interpolated Kneser-Ney and modified Kneser-Ney smoothing. However, estimation of Bayesian language models is expensive in terms of both computation time and memory; the inference is approximate and requires a number of iterations to converge. In this paper, we present a parallel training algorithm for the HPYLM, which enables the approach to be applied in the context of automatic speech recognition, using large training corpora with large vocabularies. We demonstrate the effectiveness of the proposed algorithm by estimating language models from corpora for meeting transcription containing over 200 million words, and observe significant reductions in perplexity and word error rate.}
}
@article{McGowanBerger2009,
  author = {McGowan, Richard S. and Berger, Michael A.},
  title = {Acoustic-articulatory mapping in vowels by locally weighted regression},
  journal = {Journal of the Acoustical Society of America},
  number = {4},
  pages = {2011-2032},
  volume = {126},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/aam.pdf},
  abstract = {A method for mapping between simultaneously measured articulatory and acoustic data is proposed. The method uses principal components analysis on the articulatory and acoustic variables, and mapping between the domains by locally weighted linear regression, or loess [Cleveland, W. S. (1979) J. Am. Stat. Assoc. 74, 829--836]. The latter method permits local variation in the slopes of the linear regression, assuming that the function being approximated is smooth. The methodology is applied to vowels of four speakers in the Wisconsin X-ray Microbeam Speech Production Database, with formant analysis. Results are examined in terms of (1) examples of forward (articulation-to-acoustics) mappings and inverse mappings, (2) distributions of local slopes and constants, (3) examples of correlations among slopes and constants, (4) root-mean-square error, and (5) sensitivity of formant frequencies to articulatory change. It is shown that the results are qualitatively correct and that loess performs better than global regression. The forward mappings show different root-mean-square error properties than the inverse mappings indicating that this method is better suited for the forward mappings than the inverse mappings, at least for the data chosen for the current study. Some preliminary results on sensitivity of the first two formant frequencies to the two most important articulatory principal components are presented.},
  categories = {Articulatory inversion, locally weighted regression, X-ray microbeam, formant analysis}
}
@inproceedings{tietze:09,
  author = {Tietze, Martin I. and Winterboer, Andi and Moore, Johanna D.},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/tietze.ENLG09.pdf},
  booktitle = {Proceedings ENLG09},
  title = {The effect of linguistic devices in information presentation messages on recall and comprehension},
  categories = {discourse cues, verbal information presentation, recall, eye-tracking, Mechanical Turk},
  year = {2009}
}
@article{ling2008,
  author = {Ling, Z. and Richmond, K. and Yamagishi, J. and Wang, R.},
  volume = {17},
  doi = {10.1109/TASL.2009.2014796},
  title = {Integrating Articulatory Features into {HMM}-based Parametric Speech Synthesis},
  journal = {IEEE Transactions on Audio, Speech and Language Processing},
  number = {6},
  month = {August},
  note = {\textbf{IEEE SPS 2010 Young Author Best Paper Award}},
  pages = {1171--1185},
  key = {ling2008},
  year = {2009},
  abstract = {This paper presents an investigation of ways to integrate articulatory features into Hidden Markov Model (HMM)-based parametric speech synthesis, primarily with the aim of improving the performance of acoustic parameter generation. The joint distribution of acoustic and articulatory features is estimated during training and is then used for parameter generation at synthesis time in conjunction with a maximum-likelihood criterion. Different model structures are explored to allow the articulatory features to influence acoustic modeling: model clustering, state synchrony and cross-stream feature dependency. The results of objective evaluation show that the accuracy of acoustic parameter prediction can be improved when shared clustering and asynchronous-state model structures are adopted for combined acoustic and articulatory features. More significantly, our experiments demonstrate that modeling the dependency between these two feature streams can make speech synthesis more flexible. The characteristics of synthetic speech can be easily controlled by modifying generated articulatory features as part of the process of acoustic parameter generation.},
  categories = {Speech synthesis, articulation, HMM-based synthesis}
}
@inproceedings{child_synthesis_2009,
  author = {Watts, Oliver and Yamagishi, Junichi and King, Simon and Berkling, Kay},
  title = {{HMM} Adaptation and Voice Conversion for the Synthesis of Child Speech: A Comparison},
  booktitle = {Proc. Interspeech 2009},
  address = {Brighton, U.K.},
  month = {September},
  pages = {2627--2630},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/child_synthesis_2009.pdf},
  abstract = {This study compares two different methodologies for producing data-driven synthesis of child speech from existing systems that have been trained on the speech of adults. On one hand, an existing statistical parametric synthesiser is transformed using model adaptation techniques, informed by linguistic and prosodic knowledge, to the speaker characteristics of a child speaker. This is compared with the application of voice conversion techniques to convert the output of an existing waveform concatenation synthesiser with no explicit linguistic or prosodic knowledge. In a subjective evaluation of the similarity of synthetic speech to natural speech from the target speaker, the HMM-based systems evaluated are generally preferred, although this is at least in part due to the higher dimensional acoustic features supported by these techniques.}
}
@inproceedings{Blizzard_summary_09,
  author = {King, Simon and Karaiskos, Vasilis},
  title = {The {B}lizzard {C}hallenge 2009},
  booktitle = {Proc. Blizzard Challenge Workshop},
  address = {Edinburgh, UK},
  month = {September},
  year = {2009},
  keywords = {Blizzard},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/summary_Blizzard2009.pdf},
  abstract = {The Blizzard Challenge 2009 was the fifth annual Blizzard Challenge. As in 2008, UK English and Mandarin Chinese were the chosen languages for the 2009 Challenge. The English corpus was the same one used in 2008. The Mandarin corpus was pro- vided by iFLYTEK. As usual, participants with limited resources or limited experience in these languages had the option of using unaligned labels that were provided for both corpora and for the test sentences. An accent-specific pronunciation dictionary was also available for the English speaker. This year, the tasks were organised in the form of `hubs' and `spokes' where each hub task involved building a general-purpose voice and each spoke task involved building a voice for a specific application. A set of test sentences was released to participants, who were given a limited time in which to synthesise them and submit the synthetic speech. An online listening test was conducted to evaluate naturalness, intelligibility, degree of similarity to the original speaker and, for one of the spoke tasks, "appropriateness."},
  categories = {Blizzard Challenge, speech synthesis, evaluation, listening test}
}
@inproceedings{dongwang_icassp09,
  author = {Wang, Dong and Tejedor, Tejedor and Frankel, Joe and King, Simon},
  title = {Posterior-based confidence measures for spoken term detection},
  booktitle = {Proc. ICASSP09},
  address = {Taiwan},
  month = {April},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/posterior.pdf},
  abstract = {Confidence measures play a key role in spoken term detection (STD) tasks. The confidence measure expresses the posterior probability of the search term appearing in the detection period, given the speech. Traditional approaches are based on the acoustic and language model scores for candidate detections found using automatic speech recognition, with Bayes' rule being used to compute the desired posterior probability. In this paper, we present a novel direct posterior-based confidence measure which, instead of resorting to the Bayesian formula, calculates posterior probabilities from a multi-layer perceptron (MLP) directly. Compared with traditional Bayesian-based methods, the direct-posterior approach is conceptually and mathematically simpler. Moreover, the MLP-based model does not require assumptions to be made about the acoustic features such as their statistical distribution and the independence of static and dynamic co-efficients. Our experimental results in both English and Spanish demonstrate that the proposed direct posterior-based confidence improves STD performance.},
  categories = {Spoken term detection, confidence measure, posterior probabilities, MLP},
  page = {4889--4892}
}
@article{wolters-taccess:09,
  author = {Wolters, Maria and Georgila, Kallirroi and MacPherson, Sarah and Moore, Johanna},
  title = {Being Old Doesn't Mean Acting Old: Older Users' Interaction with Spoken Dialogue Systems},
  journal = {ACM Transactions on Accessible Computing},
  number = {1},
  pages = {1--39},
  volume = {2},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/citation.cfm},
  abstract = {Most studies on adapting voice interfaces to older users work top-down by comparing the interaction behavior of older and younger users. In contrast, we present a bottom-up approach. A statistical cluster analysis of 447 appointment scheduling dialogs between 50 older and younger users and 9 simulated spoken dialog systems revealed two main user groups, a “social” group and a “factual” group. “Factual” users adapted quickly to the systems and interacted efficiently with them. “Social” users, on the other hand, were more likely to treat the system like a human, and did not adapt their interaction style. While almost all “social” users were older, over a third of all older users belonged in the “factual” group. Cognitive abilities and gender did not predict group membership. We conclude that spoken dialog systems should adapt to users based on observed behavior, not on age.},
  categories = {spoken dialogue systems, older users, human-computer interaction}
}
@inproceedings{dongwang_interspeech09_conf,
  author = {Wang, Dong and King, Simon and Frankel, Joe and Bell, Peter},
  title = {Term-Dependent Confidence for Out-of-Vocabulary Term Detection},
  booktitle = {Proc. Interspeech},
  address = {Brighton, UK},
  month = {September},
  pages = {2139--2142},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/conf.pdf},
  abstract = {Within a spoken term detection (STD) system, the decision maker plays an important role in retrieving reliable detections. Most of the state-of-the-art STD systems make decisions based on a confidence measure that is term-independent, which poses a serious problem for out-of-vocabulary (OOV) term detection. In this paper, we study a term-dependent confidence measure based on confidence normalisation and discriminative modelling, particularly focusing on its remarkable effectiveness for detecting OOV terms. Experimental results indicate that the term-dependent confidence provides much more significant improvement for OOV terms than terms in-vocabulary.},
  categories = {joint-multigram, pronunciation model, spoken term detection, speech recognition}
}
@inproceedings{dziemianko_interspeech2009,
  author = {Dziemianko, Michal and Hofer, Gregor and Shimodaira, Hiroshi},
  title = {{HMM}-Based Automatic Eye-Blink Synthesis from Speech},
  booktitle = {Proc. Interspeech},
  address = {Brighton, UK},
  month = {September},
  pages = {1799--1802},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/dziemianko_interspeech2009.pdf},
  abstract = {In this paper we present a novel technique to automatically synthesise eye blinking from a speech signal. Animating the eyes of a talking head is important as they are a major focus of attention during interaction. The developed system predicts eye blinks from the speech signal and generates animation trajectories automatically employing a ''Trajectory Hidden Markov Model''. The evaluation of the recognition performance showed that the timing of blinking can be predicted from speech with an F-score value upwards of 52\%, which is well above chance. Additionally, a preliminary perceptual evaluation was conducted, that confirmed that adding eye blinking significantly improves the perception the character. Finally it showed that the speech synchronised synthesised blinks outperform random blinking in naturalness ratings.},
  categories = {animation, motion synthesis, time series analysis, trajectory model}
}
@inproceedings{leo_09-1,
  author = {Badino, Leonardo and Andersson, J. Sebastian and Yamagishi, Junichi and Clark, Robert A.J.},
  title = {Identification of Contrast and Its Emphatic Realization in {HMM}-based Speech Synthesis},
  booktitle = {Proc. Interspeech 2009},
  address = {Brighton, U.K.},
  month = {September},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/IS090749.PDF},
  abstract = {The work presented in this paper proposes to identify contrast in the form of contrastive word pairs and prosodically signal it with emphatic accents in a Text-to-Speech (TTS) application using a Hidden-Markov-Model (HMM) based speech synthesis system. We first describe a novel method to automatically detect contrastive word pairs using textual features only and report its performance on a corpus of spontaneous conversations in English. Subsequently we describe the set of features selected to train a HMM-based speech synthesis system and attempting to properly control prosodic prominence (including emphasis). Results from a large scale perceptual test show that in the majority of cases listeners judge emphatic contrastive word pairs as acceptable as their non-emphatic counterpart, while emphasis on non-contrastive pairs is almost never acceptable.}
}
@article{yamagishi2009,
  author = {Yamagishi, Junichi and Nose, Takashi and Zen, Heiga and Ling, Zhenhua and Toda, Tomoki and Tokuda, Keiichi and King, Simon and Renals, Steve},
  title = {Robust Speaker-Adaptive {HMM}-based Text-to-Speech Synthesis},
  url = {http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=5109758&arnumber=5153555&count=14&index=12},
  journal = {IEEE Transactions on Audio, Speech and Language Processing},
  number = {6},
  pages = {1208--1230},
  volume = {17},
  year = {2009},
  pdf = {},
  abstract = {This paper describes a speaker-adaptive HMM-based speech synthesis system. The new system, called ``HTS-2007,'' employs speaker adaptation (CSMAPLR+MAP), feature-space adaptive training, mixed-gender modeling, and full-covariance modeling using CSMAPLR transforms, in addition to several other techniques that have proved effective in our previous systems. Subjective evaluation results show that the new system generates significantly better quality synthetic speech than speaker-dependent approaches with realistic amounts of speech data, and that it bears comparison with speaker-dependent approaches even when large amounts of speech data are available. In addition, a comparison study with several speech synthesis techniques shows the new system is very robust: It is able to build voices from less-than-ideal speech data and synthesize good-quality speech even for out-of-domain sentences.}
}
@inproceedings{cabral_yrwst,
  author = {Cabral, J. and Renals, S. and Richmond, K. and Yamagishi, J.},
  title = {{HMM}-based Speech Synthesis with an Acoustic Glottal Source Model},
  booktitle = {Proc. The First Young Researchers Workshop in Speech Technology},
  month = {April},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/jscabral-yrwss2009.pdf},
  abstract = {A major cause of degradation of speech quality in HMM-based speech synthesis is the use of a simple delta pulse signal to generate the excitation of voiced speech. This paper describes a new approach to using an acoustic glottal source model in HMM-based synthesisers. The goal is to improve speech quality and parametric flexibility to better model and transform voice characteristics.},
  categories = {HMM-based Speech Synthesis, LF-Model, Glottal Spectral Separation}
}
@proceedings{DBLP:conf/iceis/2009,
  editor = {Filipe, Joaquim and Cordeiro, José},
  publisher = {Springer},
  doi = {10.1007/978-3-642-01347-8},
  isbn = {978-3-642-01346-1},
  title = {Enterprise Information Systems, 11th International Conference, ICEIS 2009, Milan, Italy, May 6-10, 2009. Proceedings},
  series = {Lecture Notes in Business Information Processing},
  booktitle = {ICEIS},
  volume = {24},
  year = {2009},
  bibsource = {DBLP, http://dblp.uni-trier.de}
}
@inproceedings{jyamagis:emime,
  author = {Yamagishi, Junichi and Lincoln, Mike and King, Simon and Dines, John and Gibson, Matthew and Tian, Jilei and Guan, Yong},
  title = {Analysis of Unsupervised and Noise-Robust Speaker-Adaptive {HMM}-Based Speech Synthesis Systems toward a Unified {ASR} and {TTS} Framework},
  booktitle = {Proc. Interspeech 2009},
  address = {Edinburgh, U.K.},
  month = {September},
  year = {2009},
  abstract = {For the 2009 Blizzard Challenge we have built an unsupervised version of the HTS-2008 speaker-adaptive HMM-based speech synthesis system for English, and a noise robust version of the systems for Mandarin. They are designed from a multidisciplinary application point of view in that we attempt to integrate the components of the TTS system with other technologies such as ASR. All the average voice models are trained exclusively from recognized, publicly available, ASR databases. Multi-pass LVCSR and confidence scores calculated from confusion network are used for the unsupervised systems, and noisy data recorded in cars or public spaces is used for the noise robust system. We believe the developed systems form solid benchmarks and provide good connections to ASR fields. This paper describes the development of the systems and reports the results and analysis of their evaluation.}
}
@inproceedings{richmond2009b,
  author = {Richmond, K.},
  title = {Preliminary Inversion Mapping Results with a New {EMA} Corpus},
  booktitle = {Proc. Interspeech},
  address = {Brighton, UK},
  month = {September},
  pages = {2835--2838},
  year = {2009},
  keywords = {acoustic-articulatory inversion mapping, neural network},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/IS090544.pdf},
  abstract = {In this paper, we apply our inversion mapping method, the trajectory mixture density network (TMDN), to a new corpus of articulatory data, recorded with a Carstens AG500 electromagnetic articulograph. This new data set, mngu0, is relatively large and phonetically rich, among other beneficial characteristics. We obtain good results, with a root mean square (RMS) error of only 0.99mm. This compares very well with our previous lowest result of 1.54mm RMS error for equivalent coils of the MOCHA fsew0 EMA data. We interpret this as showing the mngu0 data set is potentially more consistent than the fsew0 data set, and is very useful for research which calls for articulatory trajectory data. It also supports our view that the TMDN is very much suited to the inversion mapping problem.}
}
@inproceedings{richmond2009a,
  author = {Richmond, K. and Clark, R. and Fitt, S.},
  title = {Robust {LTS} rules with the {Combilex} speech technology lexicon},
  booktitle = {Proc. Interspeech},
  address = {Brighton, UK},
  month = {September},
  pages = {1295--1298},
  year = {2009},
  keywords = {combilex, letter-to-sound rules, grapheme-to-phoneme conversion},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/IS090308.pdf},
  abstract = {Combilex is a high quality pronunciation lexicon aimed at speech technology applications that has recently been released by CSTR. Combilex benefits from several advanced features. This paper evaluates one of these: the explicit alignment of phones to graphemes in a word. This alignment can help to rapidly develop robust and accurate letter-to-sound (LTS) rules, without needing to rely on automatic alignment methods. To evaluate this, we used Festival's LTS module, comparing its standard automatic alignment with Combilex's explicit alignment. Our results show using Combilex's alignment improves LTS accuracy: 86.50\% words correct as opposed to 84.49\%, with our most general form of lexicon. In addition, building LTS models is greatly accelerated, as the need to list allowed alignments is removed. Finally, loose comparison with other studies indicates Combilex is a superior quality lexicon in terms of consistency and size.}
}
@inproceedings{Ehnes2009A-Tangible-Inte,
  editor = {Bello, Lucia Lo and Iannizzotto, Giancarlo},
  author = {Ehnes, Jochen},
  isbn = {978-1-4244-3960-7},
  title = {A Tangible Interface for the {AMI} Content Linking Device -- The Automated Meeting Assistant},
  lccn = {2009900916},
  booktitle = {Proceedings of HSI 2009},
  month = {May},
  note = {Best Paper Award (Human Machine Interaction)},
  pages = {306-313},
  location = {Catania, Italy},
  year = {2009},
  keywords = {Tangible Interface, Mixed Reality, Projection System, Content Linking, Automatic Meeting Assistant},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/tt4_001902.pdf},
  abstract = {In this Paper we describe our approach to support ongoing meetings with an automated meeting assistant. The system based on the AMIDA Content Linking Device aims at providing relevant documents used in previous meetings for the ongoing meeting based on automatic speech recognition. Once the content linking device finds documents linked to a discussion about a similar subject in a previous meeting, it assumes they may be relevant for the current discussion as well. We believe that the way these documents are offered to the meeting participants is equally important as the way they are found. We developed a projection based mixed reality user interface that lets the documents appear on the table tops in front of the meeting participants. They can hand them over to others or bring them onto the shared projection screen easily if they consider them relevant for others as well. Yet, irrelevant documents do not draw too much attention from the discussion. In this paper we describe the concept and implementation of this user interface and provide some preliminary results.},
  categories = {Tangible Interface, AMI, Content Linking, Mixed Reality}
}
@phdthesis{zhang-thesis2009,
  author = {Zhang, Le},
  school = {School of Informatics, University of Edinburgh},
  title = {Modelling Speech Dynamics with Trajectory-{HMM}s},
  abstract = {The conditional independence assumption imposed by the hidden Markov models (HMMs) makes it difficult to model temporal correlation patterns in human speech. Traditionally, this limitation is circumvented by appending the first and second-order regression coefficients to the observation feature vectors. Although this leads to improved performance in recognition tasks, we argue that a straightforward use of dynamic features in HMMs will result in an inferior model, due to the incorrect handling of dynamic constraints. In this thesis I will show that an HMM can be transformed into a Trajectory-HMM capable of generating smoothed output mean trajectories, by performing a per-utterance normalisation. The resulting model can be trained by either maximising model log-likelihood or minimising mean generation errors on the training data. To combat the exponential growth of paths in searching, the idea of delayed path merging is proposed and a new time-synchronous decoding algorithm built on the concept of token-passing is designed for use in the recognition task. The Trajectory-HMM brings a new way of sharing knowledge between speech recognition and synthesis components, by tackling both problems in a coherent statistical framework. I evaluated the Trajectory-HMM on two different speech tasks using the speaker-dependent MOCHA-TIMIT database. First as a generative model to recover articulatory features from speech signal, where the Trajectory-HMM was used in a complementary way to the conventional HMM modelling techniques, within a joint Acoustic-Articulatory framework. Experiments indicate that the jointly trained acoustic-articulatory models are more accurate (having a lower Root Mean Square error) than the separately trained ones, and that Trajectory-HMM training results in greater accuracy compared with conventional Baum-Welch parameter updating. In addition, the Root Mean Square (RMS) training objective proves to be consistently better than the Maximum Likelihood objective. However, experiment of the phone recognition task shows that the MLE trained Trajectory-HMM, while retaining attractive properties of being a proper generative model, tends to favour over-smoothed trajectories among competing hypothesises, and does not perform better than a conventional HMM. We use this to build an argument that models giving a better fit on training data may suffer a reduction of discrimination by being too faithful to the training data. Finally, experiments on using triphone models show that increasing modelling detail is an effective way to leverage modelling performance with little added complexity in training.},
  month = {January},
  key = {speech recognition, speech synthesis, MOCHA, trajectory HMM},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/zhangle_thesis.pdf}
}
@article{wolters-iwc:09,
  author = {Wolters, Maria and Georgila, Kallirroi and Logie, Robert and MacPherson, Sarah and Moore, Johanna and Watson, Matt},
  title = {Reducing Working Memory Load in Spoken Dialogue Systems},
  journal = {Interacting with Computers},
  number = {4},
  pages = {276-287},
  volume = {21},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/iwc09.pdf},
  abstract = {We evaluated two strategies for alleviating working memory load for users of voice interfaces: presenting fewer options per turn and providing confirmations. Forty-eight users booked appointments using nine different dialogue systems, which varied in the number of options presented and the confirmation strategy used. Participants also performed four cognitive tests and rated the usability of each dialogue system on a standardised questionnaire. When systems presented more options per turn and avoided explicit confirmation subdialogues, both older and younger users booked appointments more quickly without compromising task success. Users with lower information processing speed were less likely to remember all relevant aspects of the appointment. Working memory span did not affect appointment recall. Older users were slightly less satisfied with the dialogue systems than younger users. We conclude that the number of options is less important than an accurate assessment of the actual cognitive demands of the task at hand.},
  categories = {spoken dialogue; ageing; older adults; cognitive aging; working memory}
}
@article{hifny2009,
  author = {Hifny, Y. and Renals, S.},
  title = {Speech Recognition Using Augmented Conditional Random Fields},
  url = {http://ieeexplore.ieee.org/xpls/abs_all.jsp?isnumber=4749447&arnumber=4749472&count=25&index=15},
  journal = {IEEE Transactions on Audio, Speech and Language Processing},
  number = {2},
  pages = {354--365},
  volume = {17},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/hifny2009.pdf},
  abstract = {Acoustic modeling based on hidden Markov models (HMMs) is employed by state-of-the-art stochastic speech recognition systems. Although HMMs are a natural choice to warp the time axis and model the temporal phenomena in the speech signal, their conditional independence properties limit their ability to model spectral phenomena well. In this paper, a new acoustic modeling paradigm based on augmented conditional random fields (ACRFs) is investigated and developed. This paradigm addresses some limitations of HMMs while maintaining many of the aspects which have made them successful. In particular, the acoustic modeling problem is reformulated in a data driven, sparse, augmented space to increase discrimination. Acoustic context modeling is explicitly integrated to handle the sequential phenomena of the speech signal. We present an efficient framework for estimating these models that ensures scalability and generality. In the TIMIT phone recognition task, a phone error rate of 23.0\% was recorded on the full test set, a significant improvement over comparable HMM-based systems.}
}
@inproceedings{john:HTSGAP,
  author = {Dines, J. and Yamagishi, J. and King, S.},
  title = {Measuring the gap between {HMM}-based {ASR} and {TTS}},
  booktitle = {Proc. Interspeech},
  address = {Brighton, U.K.},
  month = {September},
  pages = {1391--1394},
  year = {2009},
  abstract = {The EMIME European project is conducting research in the development of technologies for mobile, personalised speech-to-speech translation systems. The hidden Markov model is being used as the underlying technology in both automatic speech recognition (ASR) and text-to-speech synthesis (TTS) components, thus, the investigation of unified statistical modelling approaches has become an implicit goal of our research. As one of the first steps towards this goal, we have been investigating commonalities and differences between HMM-based ASR and TTS. In this paper we present results and analysis of a series of experiments that have been conducted on English ASR and TTS systems, measuring their performance with respect to phone set and lexicon, acoustic feature type and dimensionality and HMM topology. Our results show that, although the fundamental statistical model may be essentially the same, optimal ASR and TTS performance often demands diametrically opposed system designs. This represents a major challenge to be addressed in the investigation of such unified modelling approaches.}
}
@inproceedings{dongwang_interspeech09_cmb,
  author = {Tejedor, Javier and Wang, Dong and King, Simon and Frankel, Joe and Colas, Jose},
  title = {A Posterior Probability-based System Hybridisation and Combination for Spoken Term Detection},
  booktitle = {Proc. Interspeech},
  address = {Brighton, UK},
  month = {September},
  pages = {2131--2134},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/cmb.pdf},
  abstract = {Spoken term detection (STD) is a fundamental task for multimedia information retrieval. To improve the detection performance, we have presented a direct posterior-based confidence measure generated from a neural network. In this paper, we propose a detection-independent confidence estimation based on the direct posterior confidence measure, in which the decision making is totally separated from the term detection. Based on this idea, we first present a hybrid system which conducts the term detection and confidence estimation based on different sub-word units, and then propose a combination method which merges detections from heterogeneous term detectors based on the direct posterior-based confidence. Experimental results demonstrated that the proposed methods improved system performance considerably for both English and Spanish.},
  categories = {joint-multigram, pronunciation model, spoken term detection, speech recognition}
}
@inproceedings{bell_king_full_covariance_asru2009,
  author = {Bell, Peter and King, Simon},
  doi = {10.1109/ASRU.2009.5373344},
  title = {Diagonal Priors for Full Covariance Speech Recognition},
  booktitle = {Proc. IEEE Workshop on Automatic Speech Recognition and Understanding},
  address = {Merano, Italy},
  month = {December},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/shrinkage_asru2009.pdf},
  abstract = {We investigate the use of full covariance Gaussians for large-vocabulary speech recognition. The large number of parameters gives high modelling power, but when training data is limited, the standard sample covariance matrix is often poorly conditioned, and has high variance. We explain how these problems may be solved by the use of a diagonal covariance smoothing prior, and relate this to the shrinkage estimator, for which the optimal shrinkage parameter may itself be estimated from the training data. We also compare the use of generatively and discriminatively trained priors. Results are presented on a large vocabulary conversational telephone speech recognition task.}
}
@inproceedings{steiner_is2009a,
  author = {Steiner, I. and Richmond, K.},
  title = {Towards Unsupervised Articulatory Resynthesis of {G}erman Utterances using {EMA} data},
  booktitle = {Proc. Interspeech},
  address = {Brighton, UK},
  month = {September},
  pages = {2055--2058},
  year = {2009},
  keywords = {articulatory speech synthesis, copy synthesis, electromagnetic articulography, EMA},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/IS090558.pdf},
  abstract = {As part of ongoing research towards integrating an articulatory synthesizer into a text-to-speech (TTS) framework, a corpus of German utterances recorded with electromagnetic articulography (EMA) is resynthesized to provide training data for statistical models. The resynthesis is based on a measure of similarity between the original and resynthesized EMA trajectories, weighted by articulatory relevance. Preliminary results are discussed and future work outlined.}
}
@inproceedings{jyamagis:1000sHTS,
  author = {Yamagishi, J. and Usabaev, Bela and King, Simon and Watts, Oliver and Dines, John and Tian, Jilei and Hu, Rile and Guan, Yong and Oura, Keiichiro and Tokuda, Keiichi and Karhila, Reima and Kurimo, Mikko},
  title = {Thousands of voices for {HMM}-based speech synthesis},
  booktitle = {Proc. Interspeech},
  address = {Brighton, U.K.},
  month = {September},
  pages = {420--423},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/index.php},
  abstract = {Our recent experiments with HMM-based speech synthesis systems have demonstrated that speaker-adaptive HMM-based speech synthesis (which uses an ‘average voice model’ plus model adaptation) is robust to non-ideal speech data that are recorded under various conditions and with varying microphones, that are not perfectly clean, and/or that lack of phonetic balance. This enables us consider building high-quality voices on ’non-TTS’ corpora such as ASR corpora. Since ASR corpora generally include a large number of speakers, this leads to the possibility of producing an enormous number of voices automatically. In this paper we show thousands of voices for HMM-based speech synthesis that we have made from several popular ASR corpora such as the Wall Street Journal databases (WSJ0/WSJ1/WSJCAM0), Resource Management, Globalphone and Speecon. We report some perceptual evaluation results and outline the outstanding issues.}
}
@inproceedings{NiekraszMoore09,
  author = {Niekrasz, John and Moore, Johanna},
  title = {Participant Subjectivity and Involvement as a Basis for Discourse Segmentation},
  booktitle = {{Proceedings of the SIGDIAL 2009 Conference}},
  abstract = {We propose a framework for analyzing episodic conversational activities in terms of expressed relationships between the participants and utterance content. We test the hypothesis that linguistic features which express such properties, e.g. tense, aspect, and person deixis, are a useful basis for automatic intentional discourse segmentation. We present a novel algorithm and test our hypothesis on a set of intentionally segmented conversational monologues. Our algorithm performs better than a simple baseline and as well as or better than well-known lexical-semantic segmentation methods.},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/NiekraszMoore09.pdf},
  pages = {54--61}
}
@inproceedings{benyoussef:IS2009,
  author = {Ben Youssef, Atef and Badin, Pierre and Bailly, Gérard and Heracleous, Panikos},
  title = {Acoustic-to-articulatory inversion using speech recognition and trajectory formation based on phoneme hidden Markov models},
  booktitle = {Proc. Interspeech},
  address = {Brighton, UK},
  month = {September},
  pages = {2255-2258},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/BenYoussef_Badin_Bailly_Heracleous_HMMinversion_Interspeech_2009.pdf},
  abstract = {In order to recover the movements of usually hidden articulators such as tongue or velum, we have developed a data-based speech inversion method. HMMs are trained, in a multistream framework, from two synchronous streams: articulatory movements measured by EMA, and MFCC + energy from the speech signal. A speech recognition procedure based on the acoustic part of the HMMs delivers the chain of phonemes and together with their durations, information that is subsequently used by a trajectory formation procedure based on the articulatory part of the HMMs to synthesise the articulatory movements. The RMS reconstruction error ranged between 1.1 and 2. mm.}
}
@inproceedings{benyoussef:RJCP2009,
  author = {Ben Youssef, Atef and Tran, Viet-Anh and Badin, Pierre and Bailly, Gérard},
  title = {HMMs and GMMs based methods in acoustic-to-articulatory speech inversion},
  booktitle = {Proc. RJCP},
  address = {Avignon, France},
  year = {2009},
  pdf = {http://www.cstr.inf.ed.ac.uk/downloads/publications/2009/BenYoussef-et-al_RJCP-2009.pdf},
  pages = {186-192}
}