Improving generalisation to new speakers in spoken dialogue state tracking

Users with disabilities can greatly benefit from personalised voice-enabled environmental-control interfaces, but for users with speech impairments (e.g. dysarthria) poor ASR performance poses a challenge to successful dialogue. Statistical dialogue management has shown resilience against high ASR error rates, hence making it useful to improve the performance of these interfaces. However, little research was devoted to dialogue management personalisation to specific users so far. Recently, data driven discriminative models have been shown to yield the best performance in dialogue state tracking (the inference of the user goal from the dialogue history). However, due to the unique characteristics of each speaker, training a system for a new user when user specific data is not available can be challenging due to the mismatch between training and working conditions. This work investigates two methods to improve the performance with new speakers of a LSTM-based personalised state tracker: The use of speaker specific acoustic and ASRrelated features; and dropout regularisation. It is shown that in an environmental control system for dysarthric speakers, the combination of both techniques yields improvements of 3.5% absolute in state tracking accuracy. Further analysis explores the effect of using different amounts of speaker specific data to train the tracking system

Personalised Dialogue Management for Users with Speech Disorders

Many electronic devices are beginning to include Voice User Interfaces (VUIs) as an alternative to conventional interfaces. VUIs are especially useful for users with restricted upper limb mobility, because they cannot use keyboards and mice. These users, however, often suffer from speech disorders (e.g. dysarthria), making Automatic Speech Recognition (ASR) challenging, thus degrading the performance of the VUI. Partially Observable Markov Decision Process (POMDP) based Dialogue Management (DM) has been shown to improve the interaction performance in challenging ASR environments, but most of the research in this area has focused on Spoken Dialogue Systems (SDSs) developed to provide information, where the users interact with the system only a few times. In contrast, most VUIs are likely to be used by a single speaker over a long period of time, but very little research has been carried out on adaptation of DM models to specific speakers. This thesis explores methods to adapt DM models (in particular dialogue state tracking models and policy models) to a specific user during a longitudinal interaction. The main differences between personalised VUIs and typical SDSs are identified and studied. Then, state-of-the-art DM models are modified to be used in scenarios which are unique to long-term personalised VUIs, such as personalised models initialised with data from different speakers or scenarios where the dialogue environment (e.g. the ASR) changes over time. In addition, several speaker and environment related features are shown to be useful to improve the interaction performance. This study is done in the context of homeService, a VUI developed to help users with dysarthria to control their home devices. The study shows that personalisation of the POMDP-DM framework can greatly improve the performance of these interfaces

Construction Repetition Reduces Information Rate in Dialogue

Speakers repeat constructions frequently in dialogue. Due to their peculiar information-theoretic properties, repetitions can be thought of as a strategy for cost-effective communication. In this study, we focus on the repetition of lexicalised constructions—i.e., recurring multi-word units—in English open-domain spoken dialogues. We hypothesise that speakers use construction repetition to mitigate information rate, leading to an overall decrease in utterance information content over the course of a dialogue. We conduct a quantitative analysis, measuring the information content of constructions and that of their containing utterances, estimating information content with an adaptive neural language model. We observe that construction usage lowers the information content of utterances. This facilitating effect (i) increases throughout dialogues, (ii) is boosted by repetition, (iii) grows as a function of repetition frequency and density, and (iv) is stronger for repetitions of referential constructions

Neural models of language use:Studies of language comprehension and production in context

Artificial neural network models of language are mostly known and appreciated today for providing a backbone for formidable AI technologies. This thesis takes a different perspective. Through a series of studies on language comprehension and production, it investigates whether artificial neural networks—beyond being useful in countless AI applications—can serve as accurate computational simulations of human language use, and thus as a new core methodology for the language sciences

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

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

On automatic emotion classification using acoustic features

In this thesis, we describe extensive experiments on the classification of emotions from speech using acoustic features. This area of research has important applications in human computer interaction. We have thoroughly reviewed the current literature and present our results on some of the contemporary emotional speech databases. The principal focus is on creating a large set of acoustic features, descriptive of different emotional states and finding methods for selecting a subset of best performing features by using feature selection methods. In this thesis we have looked at several traditional feature selection methods and propose a novel scheme which employs a preferential Borda voting strategy for ranking features. The comparative results show that our proposed scheme can strike a balance between accurate but computationally intensive wrapper methods and less accurate but computationally less intensive filter methods for feature selection. By using the selected features, several schemes for extending the binary classifiers to multiclass classification are tested. Some of these classifiers form serial combinations of binary classifiers while others use a hierarchical structure to perform this task. We describe a new hierarchical classification scheme, which we call Data-Driven Dimensional Emotion Classification (3DEC), whose decision hierarchy is based on non-metric multidimensional scaling (NMDS) of the data. This method of creating a hierarchical structure for the classification of emotion classes gives significant improvements over other methods tested. The NMDS representation of emotional speech data can be interpreted in terms of the well-known valence-arousal model of emotion. We find that this model does not givea particularly good fit to the data: although the arousal dimension can be identified easily, valence is not well represented in the transformed data. From the recognitionresults on these two dimensions, we conclude that valence and arousal dimensions are not orthogonal to each other. In the last part of this thesis, we deal with the very difficult but important topic of improving the generalisation capabilities of speech emotion recognition (SER) systems over different speakers and recording environments. This topic has been generally overlooked in the current research in this area. First we try the traditional methods used in automatic speech recognition (ASR) systems for improving the generalisation of SER in intra– and inter–database emotion classification. These traditional methods do improve the average accuracy of the emotion classifier. In this thesis, we identify these differences in the training and test data, due to speakers and acoustic environments, as a covariate shift. This shift is minimised by using importance weighting algorithms from the emerging field of transfer learning to guide the learning algorithm towards that training data which gives better representation of testing data. Our results show that importance weighting algorithms can be used to minimise the differences between the training and testing data. We also test the effectiveness of importance weighting algorithms on inter–database and cross-lingual emotion recognition. From these results, we draw conclusions about the universal nature of emotions across different languages