4 research outputs found
Self-Supervised Vision-Based Detection of the Active Speaker as Support for Socially-Aware Language Acquisition
This paper presents a self-supervised method for visual detection of the
active speaker in a multi-person spoken interaction scenario. Active speaker
detection is a fundamental prerequisite for any artificial cognitive system
attempting to acquire language in social settings. The proposed method is
intended to complement the acoustic detection of the active speaker, thus
improving the system robustness in noisy conditions. The method can detect an
arbitrary number of possibly overlapping active speakers based exclusively on
visual information about their face. Furthermore, the method does not rely on
external annotations, thus complying with cognitive development. Instead, the
method uses information from the auditory modality to support learning in the
visual domain. This paper reports an extensive evaluation of the proposed
method using a large multi-person face-to-face interaction dataset. The results
show good performance in a speaker dependent setting. However, in a speaker
independent setting the proposed method yields a significantly lower
performance. We believe that the proposed method represents an essential
component of any artificial cognitive system or robotic platform engaging in
social interactions.Comment: 10 pages, IEEE Transactions on Cognitive and Developmental System
Joke-o-mat HD: browsing sitcoms with human derived transcripts
ABSTRACT Joke-o-mat HD is a system that allows a user to navigate sitcoms (such as Seinfeld ) by "narrative themes", including scenes, punchlines, and dialog segments. The themes can be filtered by the main actors and by keyword. For example, the user can select to see only punchlines by Kramer that contain the word "armoire". The system infers the narrative themes using segmentation of the audio track into laughter, actors, words, and music. The segmentation can be generated either by an expert annotator, via automatic methods, or by exploiting human derived (HD) "found" data such as fan-generated scripts and closed captions. We demonstrate browsing one episode of Seinfeld using all three methods of generating segmentations
New insights into hierarchical clustering and linguistic normalization for speaker diarization
Face au volume croissant de données audio et multimédia, les technologies liées à l'indexation de données et à l'analyse de contenu ont suscité beaucoup d'intérêt dans la communauté scientifique. Parmi celles-ci, la segmentation et le regroupement en locuteurs, répondant ainsi à la question 'Qui parle quand ?' a émergé comme une technique de pointe dans la communauté de traitement de la parole. D'importants progrès ont été réalisés dans le domaine ces dernières années principalement menés par les évaluations internationales du NIST. Tout au long de ces évaluations, deux approches se sont démarquées : l'une est bottom-up et l'autre top-down. L'ensemble des systèmes les plus performants ces dernières années furent essentiellement des systèmes types bottom-up, cependant nous expliquons dans cette thèse que l'approche top-down comporte elle aussi certains avantages. En effet, dans un premier temps, nous montrons qu'après avoir introduit une nouvelle composante de purification des clusters dans l'approche top-down, nous obtenons des performances comparables à celles de l'approche bottom-up. De plus, en étudiant en détails les deux types d'approches nous montrons que celles-ci se comportent différemment face à la discrimination des locuteurs et la robustesse face à la composante lexicale. Ces différences sont alors exploitées au travers d'un nouveau système combinant les deux approches. Enfin, nous présentons une nouvelle technologie capable de limiter l'influence de la composante lexicale, source potentielle d'artefacts dans le regroupement et la segmentation en locuteurs. Notre nouvelle approche se nomme Phone Adaptive Training par analogie au Speaker Adaptive TrainingThe ever-expanding volume of available audio and multimedia data has elevated technologies related to content indexing and structuring to the forefront of research. Speaker diarization, commonly referred to as the who spoke when?' task, is one such example and has emerged as a prominent, core enabling technology in the wider speech processing research community. Speaker diarization involves the detection of speaker turns within an audio document (segmentation) and the grouping together of all same-speaker segments (clustering). Much progress has been made in the field over recent years partly spearheaded by the NIST Rich Transcription evaluations focus on meeting domain, in the proceedings of which are found two general approaches: top-down and bottom-up. Even though the best performing systems over recent years have all been bottom-up approaches we show in this thesis that the top-down approach is not without significant merit. Indeed we first introduce a new purification component leading to competitive performance to the bottom-up approach. Moreover, while investigating the two diarization approaches more thoroughly we show that they behave differently in discriminating between individual speakers and in normalizing unwanted acoustic variation, i.e.\ that which does not pertain to different speakers. This difference of behaviours leads to a new top-down/bottom-up system combination outperforming the respective baseline system. Finally, we introduce a new technology able to limit the influence of linguistic effects, responsible for biasing the convergence of the diarization system. Our novel approach is referred to as Phone Adaptive Training (PAT).PARIS-Télécom ParisTech (751132302) / SudocSudocFranceF
Visual Speaker Localization Aided by Acoustic Models
The following paper presents a novel audio-visual approach for unsupervised speaker locationing. Using recordings from a single, low-resolution room overview camera and a single far-field microphone, a state-of-the art audio-only speaker localization system (traditionally called speaker diarization) is extended so that both acoustic and visual models are estimated as part of a joint unsupervised optimization problem. The speaker diarization system first automatically determines the number of speakers and estimates “who spoke when”, then, in a second step, the visual models are used to infer the location of the speakers in the video. The experiments were performed on real-world meetings using 4.5 hours of the publicly available AMI meeting corpus. The proposed system is able to exploit audio-visual integration to not only improve the accuracy of a state-of-the-art (audioonly) speaker diarization, but also adds visual speaker locationing at little incremental engineering and computation costs