Automatic Video Self Modeling for Voice Disorder

Video self modeling (VSM) is a behavioral intervention technique in which a learner models a target behavior by watching a video of him- or herself. In the field of speech language pathology, the approach of VSM has been successfully used for treatment of language in children with Autism and in individuals with fluency disorder of stuttering. Technical challenges remain in creating VSM contents that depict previously unseen behaviors. In this paper, we propose a novel system that synthesizes new video sequences for VSM treatment of patients with voice disorders. Starting with a video recording of a voice-disorder patient, the proposed system replaces the coarse speech with a clean, healthier speech that bears resemblance to the patient’s original voice. The replacement speech is synthesized using either a text-to-speech engine or selecting from a database of clean speeches based on a voice similarity metric. To realign the replacement speech with the original video, a novel audiovisual algorithm that combines audio segmentation with lip-state detection is proposed to identify corresponding time markers in the audio and video tracks. Lip synchronization is then accomplished by using an adaptive video re-sampling scheme that minimizes the amount of motion jitter and preserves the spatial sharpness. Results of both objective measurements and subjective evaluations on a dataset with 31 subjects demonstrate the effectiveness of the proposed techniques

Computational Multimedia for Video Self Modeling

Video self modeling (VSM) is a behavioral intervention technique in which a learner models a target behavior by watching a video of oneself. This is the idea behind the psychological theory of self-efficacy - you can learn or model to perform certain tasks because you see yourself doing it, which provides the most ideal form of behavior modeling. The effectiveness of VSM has been demonstrated for many different types of disabilities and behavioral problems ranging from stuttering, inappropriate social behaviors, autism, selective mutism to sports training. However, there is an inherent difficulty associated with the production of VSM material. Prolonged and persistent video recording is required to capture the rare, if not existed at all, snippets that can be used to string together in forming novel video sequences of the target skill. To solve this problem, in this dissertation, we use computational multimedia techniques to facilitate the creation of synthetic visual content for self-modeling that can be used by a learner and his/her therapist with a minimum amount of training data. There are three major technical contributions in my research. First, I developed an Adaptive Video Re-sampling algorithm to synthesize realistic lip-synchronized video with minimal motion jitter. Second, to denoise and complete the depth map captured by structure-light sensing systems, I introduced a layer based probabilistic model to account for various types of uncertainties in the depth measurement. Third, I developed a simple and robust bundle-adjustment based framework for calibrating a network of multiple wide baseline RGB and depth cameras

The benefits of acoustic perceptual information for speech processing systems

The frame-synchronized framework has dominated many speech processing systems, such as ASR and AED targeting human speech activities. These systems have little consideration for the science behind speech and treat the task as a simple statistical classification. The framework also assumes each feature vector to be equally important to the task. However, through some preliminary experiments, this study has found evidence that some concepts defined in speech perception theories such as auditory roughness and acoustic landmarks can act as heuristics to these systems and benefit them in multiple ways. Findings of acoustic landmarks hint that the idea of treating each frame equally might not be optimal. In some cases, landmark information can improve system accuracy through highlighting the more significant frames, or improve the acoustic model accuracy by training through MTL. Further investigation into the topic found experimental evidence suggesting that acoustic landmark information can also benefit end-to-end acoustic models trained through CTC loss. With the help of acoustic landmarks, CTC models can converge with less training data and achieve lower error rate. For the first time, positive results were collected on a mid-size ASR corpus (WSJ) for acoustic landmarks. The results indicate that audio perception information can benefit a broad range of audio processing systems

Dealing with linguistic mismatches for automatic speech recognition

Recent breakthroughs in automatic speech recognition (ASR) have resulted in a word error rate (WER) on par with human transcribers on the English Switchboard benchmark. However, dealing with linguistic mismatches between the training and testing data is still a significant challenge that remains unsolved. Under the monolingual environment, it is well-known that the performance of ASR systems degrades significantly when presented with the speech from speakers with different accents, dialects, and speaking styles than those encountered during system training. Under the multi-lingual environment, ASR systems trained on a source language achieve even worse performance when tested on another target language because of mismatches in terms of the number of phonemes, lexical ambiguity, and power of phonotactic constraints provided by phone-level n-grams. In order to address the issues of linguistic mismatches for current ASR systems, my dissertation investigates both knowledge-gnostic and knowledge-agnostic solutions. In the first part, classic theories relevant to acoustics and articulatory phonetics that present capability of being transferred across a dialect continuum from local dialects to another standardized language are re-visited. Experiments demonstrate the potentials that acoustic correlates in the vicinity of landmarks could help to build a bridge for dealing with mismatches across difference local or global varieties in a dialect continuum. In the second part, we design an end-to-end acoustic modeling approach based on connectionist temporal classification loss and propose to link the training of acoustics and accent altogether in a manner similar to the learning process in human speech perception. This joint model not only performed well on ASR with multiple accents but also boosted accuracies of accent identification task in comparison to separately-trained models

Untersuchungen der rhythmischen Struktur von Sprache unter Alkoholeinfluss

This thesis is concerned with the rhythmical structure of speech under the influence of alcohol. All analyses presented are based on the Alcohol Language Corpus, which is a collection of speech uttered by 77 female and 85 male sober and intoxicated speakers. Experimental research was carried out to find robust, automatically extractable features of the speech signal that indicate speaker intoxication. These features included rhythm measures, which reflect the durational variability of vocalic and consonantal elements and are normally used to classify languages into different rhythm classes. The durational variability was found to be greater in the speech of intoxicated individuals than in the speech of sober individuals, which suggests, that speech of intoxicated speakers is more irregular than speech of sober speakers. Another set of features describes the dynamics of the short-time energy function of speech. Therefore different measures are derived from a sequence of energy minima and maxima. The results also reveal a greater irregularity in the speech of intoxicated individuals. A separate investigation about speaking rate included two different measures. One is based on the phonetic segmentation and is an estimate of the number of syllables per second. The other is the mean duration of the time intervals between successive maxima of the short-time energy function of speech. Both measures denote a decreased speaking rate in the speech of intoxicated speakers compared to speech uttered in sober condition. The results of a perception experiment show that a decrease in speaking rate also is an indicator for intoxication in the perception of speech. The last experiment investigates rhythmical features based on the fundamental frequency and energy contours of speech signals. Contours are compared directly with different distance measures (root mean square error, statistical correlation and the Euclidean distance in the spectral space of the contours). They are also compared by parameterization of the contours using Discrete Cosine Transform and the first and second moments of the lower DCT spectrum. A Principal Components Analysis on the contour data was also carried out to find fundamental contour forms regarding the speech of intoxicated and sober individuals. Concerning the distance measures, contours of speech signals uttered by intoxicated speakers differ significantly from contours of speech signals uttered in sober condition. Parameterization of the contours showed that fundamental frequency contours of speech signals uttered by intoxicated speakers consist of faster movements and energy contours of speech signals uttered by intoxicated speakers of slower movements than the respective contours of speech signals uttered in sober condition. Principal Components Analysis did not find any interpretable fundamental contour forms that could help distinguishing contours of speech signals of intoxicated speakers from those of speech uttered in sober condition. All analyses prove that the effects of alcoholic intoxication on different features of speech cannot be generalized but are to a great extent speaker-dependent

Unsupervised learning for text-to-speech synthesis

This thesis introduces a general method for incorporating the distributional analysis of textual and linguistic objects into text-to-speech (TTS) conversion systems. Conventional TTS conversion uses intermediate layers of representation to bridge the gap between text and speech. Collecting the annotated data needed to produce these intermediate layers is a far from trivial task, possibly prohibitively so for languages in which no such resources are in existence. Distributional analysis, in contrast, proceeds in an unsupervised manner, and so enables the creation of systems using textual data that are not annotated. The method therefore aids the building of systems for languages in which conventional linguistic resources are scarce, but is not restricted to these languages. The distributional analysis proposed here places the textual objects analysed in a continuous-valued space, rather than specifying a hard categorisation of those objects. This space is then partitioned during the training of acoustic models for synthesis, so that the models generalise over objects' surface forms in a way that is acoustically relevant. The method is applied to three levels of textual analysis: to the characterisation of sub-syllabic units, word units and utterances. Entire systems for three languages (English, Finnish and Romanian) are built with no reliance on manually labelled data or language-specific expertise. Results of a subjective evaluation are presented

Analyse acoustique de la voix émotionnelle de locuteurs lors d'une interaction humain-robot

Mes travaux de thèse s'intéressent à la voix émotionnelle dans un contexte d'interaction humain-robot. Dans une interaction réaliste, nous définissons au moins quatre grands types de variabilités : l'environnement (salle, microphone); le locuteur, ses caractéristiques physiques (genre, âge, type de voix) et sa personnalité; ses états émotionnels; et enfin le type d'interaction (jeu, situation d'urgence ou de vie quotidienne). A partir de signaux audio collectés dans différentes conditions, nous avons cherché, grâce à des descripteurs acoustiques, à imbriquer la caractérisation d'un locuteur et de son état émotionnel en prenant en compte ces variabilités.Déterminer quels descripteurs sont essentiels et quels sont ceux à éviter est un défi complexe puisqu'il nécessite de travailler sur un grand nombre de variabilités et donc d'avoir à sa disposition des corpus riches et variés. Les principaux résultats portent à la fois sur la collecte et l'annotation de corpus émotionnels réalistes avec des locuteurs variés (enfants, adultes, personnes âgées), dans plusieurs environnements, et sur la robustesse de descripteurs acoustiques suivant ces quatre variabilités. Deux résultats intéressants découlent de cette analyse acoustique: la caractérisation sonore d'un corpus et l'établissement d'une liste "noire" de descripteurs très variables. Les émotions ne sont qu'une partie des indices paralinguistiques supportés par le signal audio, la personnalité et le stress dans la voix ont également été étudiés. Nous avons également mis en oeuvre un module de reconnaissance automatique des émotions et de caractérisation du locuteur qui a été testé au cours d'interactions humain-robot réalistes. Une réflexion éthique a été menée sur ces travaux.This thesis deals with emotional voices during a human-robot interaction. In a natural interaction, we define at least, four kinds of variabilities: environment (room, microphone); speaker, its physic characteristics (gender, age, voice type) and personality; emotional states; and finally the kind of interaction (game scenario, emergency, everyday life). From audio signals collected in different conditions, we tried to find out, with acoustic features, to overlap speaker and his emotional state characterisation taking into account these variabilities.To find which features are essential and which are to avoid is hard challenge because it needs to work with a high number of variabilities and then to have riche and diverse data to our disposal. The main results are about the collection and the annotation of natural emotional corpora that have been recorded with different kinds of speakers (children, adults, elderly people) in various environments, and about how reliable are acoustic features across the four variabilities. This analysis led to two interesting aspects: the audio characterisation of a corpus and the drawing of a black list of features which vary a lot. Emotions are ust a part of paralinguistic features that are supported by the audio channel, other paralinguistic features have been studied such as personality and stress in the voice. We have also built automatic emotion recognition and speaker characterisation module that we have tested during realistic interactions. An ethic discussion have been driven on our work.PARIS11-SCD-Bib. électronique (914719901) / SudocSudocFranceF