88 research outputs found

    A collaborative web platform for sound archives management and analysis

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    In the context of digital sound archives, an innovative web framework for automatic analysis and manual annotation of audio files has been developed. This web framework, is called Timeside and is available under an open-source license. The TimeSide framework associates an audio processing engine, an audio database, a web API and a client-side multimedia player. The audio processing engine is written in Python language and has been designed for speech and audio signal analysis and Music Information Retrieval (MIR) tasks. It includes a set of audio analysis plugins and additionally wraps several state-of-the-art audio features extraction libraries to provide automatic annotation, segmentation and Music Information Retrieval analysis. It also provides decoding and encoding methods for most common multimedia formats. The audio database application is handled through Django (Python) and is interfaced with the audio processing engine. The web API component provides these functionalities over the web to enable web client to run analysis on the sounds in the audio database. Last but not least, the multimedia player provides an web player associated with several sound and analysis visualizations together with an annotations editor through a multi-tracks display. The TimeSide platform is available as an open-source project at the following addresses: TimeSide: https://github.com/Parisson/TimeSid

    Annulation d'écho acoustique dans les prothèses auditives

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    - Afin de pallier le phénomène d'écho au sein des prothèses auditives, l'utilisation d'une version du Block-LMS dans le domaine de la transformée de Hartley est proposée ici. La méthode de filtrage adaptatif repose sur une décomposition à délais multiples du filtre et sur une nouvelle approche dans la gestion du pas d'adaptation. Dans le but de vérifier le bon comportement de l'algorithme dans des conditions réalistes, une plate-forme de simulation en temps réel a été mise en place

    WASABI: a Two Million Song Database Project with Audio and Cultural Metadata plus WebAudio enhanced Client Applications

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    This paper presents the WASABI project, started in 2017, which aims at (1) the construction of a 2 million song knowledge base that combines metadata collected from music databases on the Web, metadata resulting from the analysis of song lyrics, and metadata resulting from the audio analysis, and (2) the development of semantic applications with high added value to exploit this semantic database. A preliminary version of the WASABI database is already online1 and will be enriched all along the project. The main originality of this project is the collaboration between the algorithms that will extract semantic metadata from the web and from song lyrics with the algorithms that will work on the audio. The following WebAudio enhanced applications will be associated with each song in the database: an online mixing table, guitar amp simulations with a virtual pedal-board, audio analysis visualization tools, annotation tools, a similarity search tool that works by uploading audio extracts or playing some melody using a MIDI device are planned as companions for the WASABI database

    Surface water temperature, salinity, and density changes in the northeast Atlantic during the last 45,000 years: Heinrich events, deep water formation, and climatic rebounds

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    We developed a new method to calculate sea surface salinities (SSS) and densities (SSD) from planktonic foraminiferal delta(18)O and sea surface temperatures (SST) as determined from planktonic foraminiferal species abundances. SST, SSS, and SSD records were calculated for the last 45,000 years for Biogeochemical Oceanic Flux Study (BOFS) cores 5K and 8K recovered from the northeast Atlantic. The strongest feature is the dramatic drop in all three parameters during the Heinrich ''ice-rafting'' events. We modelled the possibility of deepwater formation in the northeast Atlantic from the SSD records, by assuming that the surface waters at our sites cooled as they flowed further north. Comparison with modelled North Atlantic deepwater densities indicates that there could have been periods of deepwater formation between 45,000 and 30,000 C-14 years B.P. (interrupted by iceberg meltwater input of Heinrich event 3 and 4, at 27,000 and 38,000 C-14 years B.P.) and during the Holocene. No amount of cooling in the northeast Atlantic between 30,000 and 13,000 years could cause deep water to form, because of the low salinities resulting from the high meltwater inputs from icebergs. Our records indicate that after each Heinrich event there were periods of climatic rebound, with milder conditions persisting for up to 2000 years, as indicated by the presence of warmer and more saline water masses. After these warm periods conditions returned to average glacial levels. These short term cold and warm episodes in the northeast Atlantic ate superimposed on the general trend towards colder conditions of the Last Glacial Maximum (LGM). Heinrich event 1 appears to be unique as it occurs as insolation rose and was coeval with the initial melting of the Fennoscandian ice sheet. We propose that meltwater input of Heinrich event 1 significantly reduced North Atlantic Deep Water formation reducing the heat exchange between the low and high latitudes, thus delaying deglaciation by about 1500 radiocarbon years (2000 calendar years)

    Zoom arrière. L’ethnomusicologie à l’ère du Big Data

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    Les techniques d’extraction automatique d’information musicale appliquées à des répertoires de musiques traditionnelles ont ouvert de nouvelles perspectives dans le champ de l’ethnomusicologie. Si les chercheurs anglophones ont adopté le terme computational ethnomusicology pour désigner cette branche spécifique de l’ethnomusicologie, une même segmentation disciplinaire n’a pas eu lieu en France où ce nouveau champ de la recherche s’inscrit plus globalement dans le domaine des humanités numériques. Cet article collectif se propose de faire un état des lieux de la recherche dans ce domaine émergent. Nous nous interrogerons sur le devenir de l’ethnomusicologie dans ce contexte de « révolution numérique » où les dispositifs informatiques bouleversent d’ores et déjà les pratiques de recherche

    A multi-modal dance corpus for research into interaction between humans in virtual environments

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    We present a new, freely available, multimodal corpus for research into, amongst other areas, real-time realistic interaction between humans in online virtual environments. The specific corpus scenario focuses on an online dance class application scenario where students, with avatars driven by whatever 3D capture technology is locally available to them, can learn choreographies with teacher guidance in an online virtual dance studio. As the dance corpus is focused on this scenario, it consists of student/teacher dance choreographies concurrently captured at two different sites using a variety of media modalities, including synchronised audio rigs, multiple cameras, wearable inertial measurement devices and depth sensors. In the corpus, each of the several dancers performs a number of fixed choreographies, which are graded according to a number of specific evaluation criteria. In addition, ground-truth dance choreography annotations are provided. Furthermore, for unsynchronised sensor modalities, the corpus also includes distinctive events for data stream synchronisation. The total duration of the recorded content is 1 h and 40 min for each single sensor, amounting to 55 h of recordings across all sensors. Although the dance corpus is tailored specifically for an online dance class application scenario, the data is free to download and use for any research and development purposes

    Traitement numérique du signal acoustique pour une aide aux malentendants

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    Hearing impairment may be considered as a serious physical and social handicap. Alleviating hearing impairment is therefore a very challenging application for digital signal processing. The framework of this thesis is the use and development of digital signal processing technique for use in hearing-aid devices. Since the auditory system is a very effective and complex, compensating for the hearing-loss is a very challenging problem. Moreover, hearing-aid is based on miniaturized chips and electro-acoustic devices that force to be subject to certain technology constraints. During our research, we have studied and developed a speech enhancement stage in the hearing-aid using a short-term spectral modification method, the Ephra�m and Malah suppression rule (EMSR). In order to perform the frequency analysis on a perceptually relevant frequency scale, we introduce an adaptation of the EMSR. The frequency scale used in this work, the ERB (Equivalent Rectangular Bandwith) scale, models the resolution of the auditory filters. Hearing-aid can be considered as a closed-loop electro-acoustical system that can become unstable because of the feedback. The main part of this thesis is devoted to feedback cancellation using multi delay adaptive filtering in the transform domain. The last part of this work consist in developing a real-time implementation of the feedback cancellation algorithm previously proposed using a prototype hearing aid.Pour un malentendant, le déficit auditif peut représenter un handicap majeur tant du point de vue physique que du point de vue social. Pour cette raison, la réhabilitation auditive constitue un champ d'application particulièrement important du traitement du signal audio et ses enjeux dépassent le seul contexte technologique. C'est dans ce cadre, et en particulier dans celui de la réhabilitation auditive par port de prothèses externes, que se situe ce travail de thèse. L'audition étant un sens particulièrement performant et complexe, compenser les pertes auditives des malentendants est une tâche difficile et délicate. En outre, les appareils sont conçus à partir d'éléments miniaturisés de pointe, notamment les micro-processeurs, ce qui impose des contraintes de fonctionnement très strictes. Au cours de nos recherches, nous avons travaillé sur la réalisation d'un module de rehaussement du signal de parole utilisant une technique d'atténuation spectrale à court-terme mise au point par Ephraïm et Malah. A l'issue de ce travail, nous avons proposé d'adapter cette méthode en réalisant l'analyse spectrale selon une échelle en fréquence conforme à notre perception auditive, l'échelle ERB (Equivalent Rectangular Bandwith) qui modélise la résolution fréquentielle des filtres auditifs. Dans un deuxième temps, nous avons étudié le problème de retour acoustique entre les transducteurs de la prothèse. Ce retour acoustique est source d'instabilité pour le système et aboutit fréquemment à la production d'un effet Larsen. Ce type de désagrément est extrêmement gênant et contraint l'audioprothésiste à limiter le gain d'amplification de la prothèse. De ce fait, la réhabilitation auditive ne peut pas être optimale. Pour pallier ce problème, nous proposons d'utiliser une méthode de filtrage adaptatif multi-délais basée sur le LMS (Least Mean Square) et réalisée dans le domaine de la transformée de Hartley. Du fait du retour acoustique, une prothèse auditive est un système électro-acoustique en boucle fermée. Les conditions d'application sont donc contraires aux hypothèses de convergence du LMS. Pour assurer la convergence de l'algorithme de descente de gradient, nous proposons une nouvelle méthode de définition du pas d'adaptation. Afin de vérifier le bon fonctionnement de la méthode proposée dans des conditions réalistes, nous avons mis en place une plate-forme de test en temps-réel basée sur un prototype de prothèses

    Traitement numérique du signal acoustique pour une aide aux malentendants

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