Добірка наукової літератури з теми "Antennes de microphones"
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Статті в журналах з теми "Antennes de microphones":
Yon, H., N. H. Abd Rahman, M. A. Aris, and Hadi Jumaat. "Developed high gain microstrip antenna like microphone structure for 5G application." International Journal of Electrical and Computer Engineering (IJECE) 10, no. 3 (June 1, 2020): 3086. http://dx.doi.org/10.11591/ijece.v10i3.pp3086-3094.
Donavan, Paul R. "Application of Sound Intensity to the Measurement of Aeroacoustic Noise Sources in Flow." INTER-NOISE and NOISE-CON Congress and Conference Proceedings 266, no. 2 (May 25, 2023): 839–46. http://dx.doi.org/10.3397/nc_2023_01_1121.
Sezen, A. S., S. Sivaramakrishnan, S. Hur, R. Rajamani, W. Robbins, and B. J. Nelson. "Passive Wireless MEMS Microphones for Biomedical Applications." Journal of Biomechanical Engineering 127, no. 6 (July 8, 2005): 1030–34. http://dx.doi.org/10.1115/1.2049330.
Krasny, Leonid. "Speech recognition using microphone antenna array." Journal of the Acoustical Society of America 119, no. 2 (2006): 690. http://dx.doi.org/10.1121/1.2174515.
Rohde, Charles A., and Christina J. Naify. "Detecting acoustic chirality with matched metamaterial vortex wave antennas." Journal of the Acoustical Society of America 154, no. 2 (August 1, 2023): 721–29. http://dx.doi.org/10.1121/10.0020533.
Romero, Daniel, and Roberto Lopez-Valcarce. "Spectrum Sensing for Wireless Microphone Signals Using Multiple Antennas." IEEE Transactions on Vehicular Technology 63, no. 9 (November 2014): 4395–407. http://dx.doi.org/10.1109/tvt.2014.2316513.
Blanchard, Torea, Jean-Hugh Thomas, and Kosai Raoof. "Acoustic Signature Analysis for Localization Estimation of Unmanned Aerial Vehicles Using Few Number of Microphones." MATEC Web of Conferences 283 (2019): 04002. http://dx.doi.org/10.1051/matecconf/201928304002.
Wielgus, Agnieszka, and Bogusław Szlachetko. "A General Scheme of a Branch-and-Bound Approach for the Sensor Selection Problem in Near-Field Broadband Beamforming." Sensors 24, no. 2 (January 12, 2024): 470. http://dx.doi.org/10.3390/s24020470.
Caronna, Gaetano, Ivan Roselli, and Pierluigi Testa. "Modeling of phased antenna array of about 500 microphones, detecting landing airplanes." Journal of the Acoustical Society of America 120, no. 5 (November 2006): 3218–19. http://dx.doi.org/10.1121/1.4788172.
Berry, Alain. "Assessment of acoustical materials sound absorption coefficient under oblique incidence plane wave and diffuse field using a virtual source antenna." INTER-NOISE and NOISE-CON Congress and Conference Proceedings 267, no. 1 (November 5, 2023): 53–56. http://dx.doi.org/10.3397/no_2023_0014.
Дисертації з теми "Antennes de microphones":
Demontis, Hugo. "Identification de sources acoustiques complexes en milieu réverbérant par grands réseaux de microphones." Thesis, Sorbonne université, 2019. http://www.theses.fr/2019SORUS196.
Knowing the directivity pattern of an acoustic source is useful in many applications in acoustics. To experimentally estimate the spatial signature, it is common to deploy microphones partially or totally surrounding the source. The acoustic radiation is then captured in all possible directions. In this thesis, we discuss the development of a large-scale 3D microphone array. This array, named "MODO" ("Les Murs Ont Des Oreilles", or, "The Walls Have Ears"), is comprised of 1024 digital MEMS microphones, flush mounted on the walls and the ceiling of a typical shoe-box room. In order to localize the sources and identify their directivity pattern, we solve the associated inverse problem under block-sparsity constraints. The chosen method exploits the small number of sources inside the room, allowing a sparse representation of the measured sound field. We use the spherical harmonics formalism to efficiently describe the directivity of the sources and their individual contributions to the radiation pattern. The acoustic path is modelled via integration of room transfer functions, synthesized with the mirror microphone method. We validated the proposed characterization method \textit{in situ} by comparison with known directivity patterns, calibrated using a high order spherical microphone array in controlled conditions
Hua, Thanh Phong. "Adaptation mode controllers for adaptive microphone arrays." Rennes 1, 2006. http://www.theses.fr/2006REN1S136.
Mariotte, Théo. "Traitement automatique de la parole en réunion par dissémination de capteurs." Electronic Thesis or Diss., Le Mans, 2024. http://www.theses.fr/2024LEMA1001.
This thesis work focuses on automatic speech processing, and more specifically on speaker diarization. This task requires the signal to be segmented to identify events such as voice activity, overlapped speech, or speaker changes. This work tackles the scenario where the signal is recorded by a device located in the center of a group of speakers, as in meetings. These conditions lead to a degradation in signal quality due to the distance between the speakers (distant speech).To mitigate this degradation, one approach is to record the signal using a microphone array. The resulting multichannel signal provides information on the spatial distribution of the acoustic field. Two lines of research are being explored for speech segmentation using microphone arrays.The first introduces a method combining acoustic features with spatial features. We propose a new set of features based on the circular harmonics expansion. This approach improves segmentation performance under distant speech conditions while reducing the number of model parameters and improving robustness in case of change in the array geometry.The second proposes several approaches that combine channels using self-attention. Different models, inspired by an existing architecture, are developed. Combining channels also improves segmentation under distant speech conditions. Two of these approaches make feature extraction more interpretable. The proposed distant speech segmentation systems also improve speaker diarization.Channel combination shows poor robustness to changes in the array geometry during inference. To avoid this behavior, a learning procedure is proposed, which improves the robustness in case of array mismatch.Finally, we identified a gap in the public datasets available for distant multichannel automatic speech processing. An acquisition protocol is introduced to build a new dataset, integrating speaker position annotation in addition to speaker diarization.Thus, this work aims to improve the quality of multichannel distant speech segmentation. The proposed methods exploit the spatial information provided by microphone arrays while improving the robustness in case of array mismatch
Ramamonjy, Aro. "Développement de nouvelles méthodes de classification/localisation de signaux acoustiques appliquées aux véhicules aériens." Thesis, Paris, CNAM, 2019. http://www.theses.fr/2019CNAM1234/document.
This thesis deals with the development of a compact microphone array and a dedicated signal processing chain for aerialtarget recognition and direction of arrival (DOA) estimation. The suggested global approach consists in an initial detection ofa potential target, followed by a DOA estimation and tracking process, along with a refined detection, facilitated by adaptivespatial filtering. An original DOA estimation algorithm is proposed. It uses the RANSAC algorithm on real-time time-domainbroadband [100 Hz - 10 kHz] pressure and particle velocity data which are estimated using finite differences and sums ofsignals of microphone pairs with frequency-dependent inter-microphone spacings. The use of higher order finite differences, or variants of the Phase and Amplitude Gradient Estimation (PAGE) method adapted to the designed antenna, can extend its bandwidth at high frequencies. The designed compact microphone array uses 32 digital MEMS microphones, horizontally disposed over an area of 7.5 centimeters. This array geometry is suitable to the implemented algorithms for DOA estimation and spatial filtering. DOA estimation and tracking of a trajectory controlled by a spatialization sphere in the Ambisonic domain have shown an average DOA estimation error of 4 degrees. A database of flying drones acoustic signatures has been set up, with the knowledge of the drone’s position in relation to the microphone array set out by GPS measurements. Adding artificial noise to the data, and selecting acoustic features with evolutionary programming have enabled the detection of an unknown drone in an unknown soundscape within 200 meters with the JRip classifier. In order to facilitate the detection and extend its range, the initial detection stage is preceded by differential beamforming in four main directions (north, south, east, west), and the refined detection stage is preceded by MVDR beamforming informed by the target’s DOA
Massé, Pierre. "Analysis, Treatment, and Manipulation Methods for Spatial Room Impulse Responses Measured with Spherical Microphone Arrays." Electronic Thesis or Diss., Sorbonne université, 2022. http://www.theses.fr/2022SORUS079.
The use of spatial room impulse responses (SRIR) for the reproduction of three-dimensional reverberation effects through multi-channel convolution over immersive surround-sound loudspeaker systems has become commonplace within the last few years, thanks in large part to the commercial availability of various spherical microphone arrays (SMA) as well as a constant increase in computing power. This use has in turn created a demand for analysis and treatment techniques not only capable of ensuring the faithful reproduction of the measured reverberation effect, but which could also be used to control various modifications of the SRIR in a more "creative" approach, as is often encountered in the production of immersive musical performances and installations. Within this context, the principal objective of the current thesis is the definition of a complete space-time-frequency framework for the analysis, treatment, and manipulation of SRIRs. The analysis tools should lead to an in-depth model allowing for measurements to first be treated with respect to their inherent limitations (measurement conditions, background noise, etc.), as well as offering the ability to modify different characteristics of the final reverberation effect described by the SRIR. These characteristics can be either completely objective, even physical, or otherwise informed by knowledge of human auditory perception with regard to room acoustics. The theoretical work in this research project is therefore presented in two main parts. First, the underlying SRIR signal model is described, heavily inspired by the historical approaches from the fields of artificial reverberation synthesis and SMA signal processing, while at the same time (incrementally) extending both. The signal model is then used to define the analysis methods that form the core of the final framework; these focus particularly on (a) identifying the "mixing time" that defines the moment of transition between the early reflection and late reverberation regimes, (b) obtaining a space-time cartography of the early reflections, and (c) estimating the frequency- and direction-dependent properties of the late reverberation's exponential energy decay envelope. In order to account for the directional dependence of these properties, a procedure for generating directional SRIR representations (i.e. directional room impulse responses, DRIR) that guarantee the preservation of certain fundamental reverberation properties must also be defined. In the second part, the model parameters made explicit by the analysis methods are exploited in order to either treat (i.e. attempt to correct some of the inevitable limitations inherent to the SMA measurement process) or more creatively manipulate and modify the SRIR. Two treatment methods in particular are developed in this thesis: (1) a pre-analysis procedure acting directly on repeated exponential sweep method (ESM) SMA measurement signals in an attempt to simultaneously increase the resulting SRIR's signal-to-noise ratio (SNR) while reducing its vulnerability to non-stationary noise events, and (2) a post-analysis denoising technique based on replacing the SRIR's background noise floor with a resynthesized extrapolation of the late reverberation tail. The theoretical descriptions thus complete, the main analysis methods as well as the DRIR generation and the denoising treatment procedures are then subjected to a series of validation tests, wherein simulated SRIRs (or parts thereof) are used to evaluate the performance, discuss the limitations, and parameterize the implementation of the different techniques. These sub-studies allow each method to be individually verified, resulting in a comprehensive investigation into the inner workings of the analysis toolbox (as well as the denoising process). Finally, to provide a concluding overview of the complete analysis-treatment-manipulation framework, similar studies are carried out using examples of real-world [...]
Blanchard, Torea. "Caractérisation de drones en vue de leur localisation et de leur suivi à partir d’une antenne de microphones." Thesis, Le Mans, 2019. http://www.theses.fr/2019LEMA1042.
This thesis work focuses on the acoustic identification of drones in order to design an array with few microphones (up to 10) and adapted to the frequencies emitted for localizing and tracking these devices. Characterization measurements have shown the inherent harmonic structure of the signal emitted by the UAV propulsion systems. A filtering step before the localization, adapted to this type of signal, is proposed. It consists of the detection of the fundamental frequency by the HPS (Harmonic Product Spectrum) algorithm and a series of bandpass filters to preserve the useful harmonics of the signal. Two methods of localization are compared through numerical simulations and experimental measurements. The first is beamforming in the time domain. Usually used for angular source localization, it is extended for localization in 3D space. The second, called acoustic goniometry, estimates the angular position of the target as a solution to an inverse problem. A Kalman filter is then used to track the target. An experimental measurement campaign made it possible to establish a database of the displacement of a small four-engine drone for different trajectories. Data analysis showed that a small number of harmonics (3 to 6) in the signal spectrum of the source to be located is sufficient to estimate the position of a source without significant loss in accuracy relative to a location without processing. The choice of this strategy is justified for localization and tracking in the presence of several drones
Faure, Baldrik. "Caractérisation du rayonnement acoustique d'un rail à l'aide d'un réseau de microphones." Phd thesis, Université de Grenoble, 2011. http://tel.archives-ouvertes.fr/tel-00721959.
Furnon, Nicolas. "Apprentissage profond pour le rehaussement de la parole dans les antennes acoustiques ad-hoc." Electronic Thesis or Diss., Université de Lorraine, 2021. http://www.theses.fr/2021LORR0277.
More and more devices we use in our daily life are embedded with one or more microphones so that they can be voice controlled. Put together, these devices can form a so-called ad-hoc microphone array (AHMA). A speech enhancement step is often applied on the recorded signals to optimise the execution of the voice commands. To this effect, AHMAs are of high interest because of their flexible usage, their wide spatial coverage and the diversity of their recordings. However, it is challenging to exploit the potential of mbox{AHMAs} because devices that compose them may move and have a limited power and bandwidth capacity. Because of these limits, the speech enhancement solutions deployed in ``classic'' microphone arrays, relying on a fusion center and high processing loads, cannot be afforded.This thesis combines the modelling power of deep neural networks (DNNs) with the flexibility of use of AHMAs. To this end, we introduce a distributed speech enhancement system, which does not rely on a fusion center. So-called compressed signals are sent among the nodes and convey the spatial information recorded by the whole AHMA, while reducing the bandwidth requirements. DNNs are used to estimate the coefficients of a multichannel Wiener filter. We conduct an empirical analysis of this sytem, both on synthesized and real data, in order to validate its efficiency and to highlight the benefits of jointly using DNNs and distributed speech enhancement algorithms. We show that our system performs comparatively well compared with a state-of-the-art solution, while being more flexible and significantly reducing the computation cost.Besides, we develop our solution to adapt it to the typical usage conditions of mbox{AHMAs}. We study its behaviour when the number of devices in the AHMA varies. We introduce and compare a spatial attention mechanism and a self-attention mechanism. Both mechanisms make our system robust to a varying number of devices. We show that the weights of the self-attention mechanism reveal the utility of the information carried by each signal.We also analyse our system when the signals recorded by different devices are not synchronised. We propose a solution to improve its performance in such conditions by introducing a temporal attention mechanism. We show that this mechanism can help estimating the sampling time offset between the several devices of the AHMA.Lastly, we show that our system is also efficient for source separation. It can efficiently process the spatial information recorded by the whole AHMA in a typical meeting scenario and alleviate the needs of a complex DNN architecture
Pujol, Hadrien. "Antennes microphoniques intelligentes : localisation de sources acoustiques par Deep Learning." Thesis, Paris, HESAM, 2020. http://www.theses.fr/2020HESAC025.
For my PhD thesis, I propose to explore the path of supervised learning, for the task of locating acoustic sources. To do so, I have developed a new deep neural network architecture. But, to optimize the millions of learning variables of this network, a large database of examples is needed. Thus, two complementary approaches are proposed to constitute these examples. The first is to carry out numerical simulations of microphonic recordings. The second one is to place a microphone antenna in the center of a sphere of loudspeakers which allows to spatialize the sounds in 3D, and to record directly on the microphone antenna the signals emitted by this experimental 3D sound wave simulator. The neural network could thus be tested under different conditions, and its performances could be compared to those of conventional algorithms for locating acoustic sources. The results show that this approach allows a generally more precise localization, but also much faster than conventional algorithms in the literature
Guillaume, Mathieu. "Analyse et synthèse de champs sonores." Phd thesis, Télécom ParisTech, 2006. http://pastel.archives-ouvertes.fr/pastel-00002383.
Книги з теми "Antennes de microphones":
Herbordt, Wolfgang. Sound capture for human/machine interfaces: Practical aspects of microphone array signal processing. Berlin: Springer, 2005.
Herbordt, Wolfgang. Sound capture for human/machine interfaces: Practical aspects of microphone array signal processing. Berlin: Springer, 2005.
My Ham Radio Log Book V2 V9 Editions. Ham Radio Log Book: Amateur Radio Log Book - Amateur Radio Operator Station Log Book - Ham Radio Log Sheet - 111 Pages, 8,5 X11 - Paperback - Yellow Pattern Background with Microphone, Radio Antenna, Headset. Independently Published, 2019.
Частини книг з теми "Antennes de microphones":
Spence, John C. H. "Radio and Telecommunications." In Lightspeed, 159–79. Oxford University Press, 2019. http://dx.doi.org/10.1093/oso/9780198841968.003.0009.
Тези доповідей конференцій з теми "Antennes de microphones":
Donavan, Paul R. "Application of Sound Intensity to the Measurement of Aeroacoustic Noise Sources in Flow." In Noise and Vibration Conference & Exhibition. 400 Commonwealth Drive, Warrendale, PA, United States: SAE International, 2023. http://dx.doi.org/10.4271/2023-01-1121.
Tai-liang, Ju, Peng Qi-cong, and Shao Huai-zong. "Speech Source 3D Localization Focusing Algorithms Based on Microphone Array." In 2006 7th International Symposium on Antennas, Propagation & EM Theory. IEEE, 2006. http://dx.doi.org/10.1109/isape.2006.353369.
Yon, H., M. T. Ali, M. A. Aris, B. Baharom, and N. A. M. Nasir. "A New Model Microstrip Antenna like Microphone Structure for 5G Application." In 2018 IEEE International RF and Microwave Conference (RFM). IEEE, 2018. http://dx.doi.org/10.1109/rfm.2018.8846496.
Aumann, Herbert M., Travis Russell, and Nuri W. Emanetoglu. "Comparison of a small parabolic reflector for use with an acoustic and a radar microphone." In 2015 IEEE International Symposium on Antennas and Propagation & USNC/URSI National Radio Science Meeting. IEEE, 2015. http://dx.doi.org/10.1109/aps.2015.7305488.
Catur, Hilman Adritya H. B. B., and Hendri Maja Saputra. "Azimuth Estimation based on Generalized Cross Correlation Phase Transform (GCC-PHAT) Using Equilateral Triangle Microphone Array." In 2019 International Conference on Radar, Antenna, Microwave, Electronics, and Telecommunications (ICRAMET). IEEE, 2019. http://dx.doi.org/10.1109/icramet47453.2019.8980432.
Ivanenkov, A. S., and A. A. Rodionov. "Estimation of broadband signal radiated by a single source in the presence of multiple acoustic interference sources via microphone array." In 2013 IX International Conference on Antenna Theory and Techniques (ICATT). IEEE, 2013. http://dx.doi.org/10.1109/icatt.2013.6650839.
Lissek, H., H. Esfahlani, J. R. Mosig, and S. Karkar. "Development of leaky-wave antenna applications with acoustics metamaterials: From the acoustic dispersive prism to sound direction finding with a single microphone." In 2017 11th International Congress on Engineered Materials Platforms for Novel Wave Phenomena (Metamaterials). IEEE, 2017. http://dx.doi.org/10.1109/metamaterials.2017.8107890.
Kim, Jaehwan, Sang Yeol Yang, Min Hee Lee, Jung Hwan Kim, Zhijiang Cai, Joo Hyung Kim, and Kwang Sun Kang. "Cellulose Smart Material for Sensor, Actuator and MEMS Applications." In ASME 2008 Conference on Smart Materials, Adaptive Structures and Intelligent Systems. ASMEDC, 2008. http://dx.doi.org/10.1115/smasis2008-381.