10 research outputs found

    Aplicación de algoritmos combinados de filtrado adaptativo a acústica de salas

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    Las aplicaciones de procesamiento de señales acústicas están cobrando una importancia creciente. La mayoría de aplicaciones de este tipo (como la cancelación de eco acústico, la cancelación de ruido, la dereverberación, la separación y el seguimiento de fuentes acústicas, etc.) requieren la identificación de una (o varias) respuestas al impulso del recinto (RIRs). Estas respuestas pueden variar con el tiempo, por lo que se precisa de esquemas adaptativos para su identificación. La utilización de esquemas adaptativos en escenarios de identificación de respuestas acústicas se ve sujeta a diferentes compromisos, como, p. ej., la conocida relación entre velocidad de convergencia y precisión en estacionario. Varios de estos compromisos se comparten con otras aplicaciones, mientras que otros son específicos del procesamiento de señales acústicas. Entre los diferentes métodos que tratan de aliviar estas limitaciones, destaca la combinación adaptativa de filtros adaptativos debido fundamentalmente a su sencillez, versatilidad y eficacia. En esta Tesis Doctoral se aborda el estudio, diseño, implementación y adecuación de los esquemas de combinación adaptativa para que resulten provechosos y convenientes en aplicaciones de procesamiento de señales acústicas. Para ello, se proponen y analizan esquemas de combinación que ofrecen robustez y un comportamiento adecuado con respecto a las particularidades que presentan las señales acústicas involucradas y las RIRs. De entre los posibles condicionantes y sus potenciales soluciones, en esta Tesis Doctoral se contemplan: - La relación señal a ruido es normalmente desconocida a priori y puede variar. Se han desarrollado dos esquemas de combinación de filtros robustos frente a cambios en dicha relación. - El espectro de las señales acústicas (música y voz) no es plano en frecuencia, lo que ralentiza la convergencia de los filtros adaptativos. Se presenta un algoritmo de combinación en el dominio frecuencial que permite combinar de forma independiente diferentes bandas de frecuencia, obteniendo ganancias debido a que, por lo general, la relación señal a ruido es diferente en cada subbanda, y los cambios producidos en la RIR no afectan de igual forma a todo el margen frecuencial. En algunos casos, la relación entre la señal a reproducir por los altavoces y la captada por los transductores receptores es no lineal. La solución estándar para este problema de identificación no lineal se basa normalmente en los filtros de Volterra, y esta Tesis Doctoral presenta dos novedosas estrategias de combinación ad-hoc para su utilización en este contexto, las cuales obtienen ventajas de las particularidades de este tipo de filtros. Además, se propone un esquema que presenta una gran robustez con respecto a la ausencia o presencia de distorsión no lineal, e incluso con respecto a variaciones en la potencia de esta distorsión, con un modesto incremento de coste computacional con respecto al de un filtro de Volterra clásico. En muchas ocasiones, la longitud de la RIR es grande y la distribución de su energía no uniforme. Se propone un esquema que, explotando el compromiso entre sesgo y varianza, permite ganancias en esta situación, principalmente cuando la relación señal a ruido es baja. Para mostrar las ventajas del uso de los esquemas de combinación propuestos, se han llevado a cabo una serie de experimentos utilizando un escenario de cancelación de eco acústico monocanal. En todos los casos, las soluciones presentadas han obtenido resultados satisfactorios, demostrando la versatilidad y el potencial de estos algoritmos, y permitiendo mejorar el funcionamiento de los filtros adaptativos ante los condicionantes anteriormente citados. ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------Acoustic signal processing applications are becoming increasingly important. Most of these applications, such as acoustic echo cancellation, noise cancellation, dereverberation, separation and tracking of acoustic sources, etc., requires the identification of a (or several) room impulse response (RIR). This response is usually time-varying, what justifies the use of adaptive algorithms to carry out the identification task. The use of adaptive schemes in RIR identification scenarios is subject to different compromises, such as the well-known compromise between speed of convergence and steady-state precision. Several of these tradeoffs are shared by other applications, while others are specific to acoustic signal processing. Among the different methods available to alleviate these limitations, adaptive combination of adaptive filters has been recently receiving a lot of attention, mainly because of its simplicity, versatility, and effectiveness. In this Ph. D. Thesis, we deal with the development, study and implementation of adaptive combination schemes that are especially suited to acoustic signal processing applications. For this purpose, we propose and analyze combination schemes that offer robustness and a suitable behavior with respect to the peculiarities of the involved signals and RIRs. Among all possible determining factors and their potential solutions, in this Ph. D. Thesis we consider: The signal to noise ratio is usually unknown a priori and it can be time-varying. In order to deal with this situation, two new different schemes are proposed. The spectrum of acoustic signals (music and speech) is not flat, what slows down the convergence of adaptive filters. We present a combination algorithm in the frequency domain that allows to mix different frequency bands independently, offering gains that exploit the frequency dependent signal to noise power ratio and the fact that RIR changes can also take place in a frequency-localized manner. Occasionally, the relationship between the signal to be reproduced by the loudspeakers and the signal received by the microphones is nonlinear. The standard solution for this nonlinear identification problem is frequently based on Volterra filters. The Thesis presents two novel ad-hoc combinations strategies to be used in this context, which take advantage of the particularities of this kind of filters. In addition, we propose an additional algorithm that shows great robustness with respect to the presence or absence of nonlinear distortion, and even with respect to changes in the power of nonlinear distortion, with a very modest increment in terms of computational cost. In many cases, very large RIRs are present, and their energies are typically distributed in a non-uniform manner. We propose a scheme that, exploiting the tradeoff between bias and variance, permits important gains in this situation, mainly for low signal to noise power ratios. In order to illustrate the advantages of the proposed combinations schemes, several experiments have been carried out considering a single-channel acoustic echo cancellation scenario. The satisfactory results obtained by the presented solutions demonstrate the versatility and potential of these algorithms, allowing to improve the performance of adaptive filters in the presence of the aforementioned conditions

    Steady-State Performance of an Adaptive Combined MISO Filter Using the Multichannel Affine Projection Algorithm

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    The combination of adaptive filters is an effective approach to improve filtering performance. In this paper, we investigate the performance of an adaptive combined scheme between two adaptive multiple-input single-output (MISO) filters, which can be easily extended to the case of multiple outputs. In order to generalize the analysis, we consider the multichannel affine projection algorithm (APA) to update the coefficients of the MISO filters, which increases the possibility of exploiting the capabilities of the filtering scheme. Using energy conservation relations, we derive a theoretical behavior of the proposed adaptive combination scheme at steady state. Such analysis entails some further theoretical insights with respect to the single channel combination scheme. Simulation results prove both the validity of the theoretical steady-state analysis and the effectiveness of the proposed combined scheme.The work of Danilo Comminiello, Michele Scarpiniti and Aurelio Uncini has been supported by the project: “Vehicular Fog energy-efficient QoS mining and dissemination of multimedia Big Data streams (V-FoG and V-Fog2)”, funded by Sapienza University of Rome Bando 2016 and 2017. The work of Michele Scarpiniti and Aurelio Uncini has been also supported by the project: “GAUChO – A Green Adaptive Fog Computing and networking Architectures” funded by the MIUR Progetti di Ricerca di Rilevante Interesse Nazionale (PRIN) Bando 2015, grant 2015YPXH4W_004. The work of Luis A. Azpicueta-Ruiz is partially supported by the Spanish Ministry of Economy and Competitiveness (under grant DAMA (TIN2015-70308-REDT) and grants TEC2014-52289-R and TEC2017-83838-R), and by the European Union

    Colección de prácticas de acústica de recintos

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    Grado en Ingeniería de Sistemas Audiovisuales. Asignatura: Acústica de recintosEl objetivo de la asignatura optativa Acústica de Recintos es complementar el contenido de la asignatura troncal Sistemas Electroacústicos y Sonorización, en dos aspectos básicos: El desarrollo de las teorías clásicas de propagación acústica que describen el comportamiento del campo sonoro en el interior de un recinto (estadística, geométrica y ondulatoria) y la profundización en los conceptos de acondicionamiento acústico y aislamiento acústico. Como parte del material de la asignatura Acústica de Recintos, este manual docente incluye los enunciados de tres prácticas a impartir durante la asignatura. Las dos primeras prácticas tienen el objetivo de que el alumno se familiarice con los softwares de simulación acústica en recintos cerrados. De esta forma, es posible reproducir el campo sonoro de recintos existentes o incluso obtener información sobre cómo sonarían recintos que no existen en la actualidad. El manejo de este tipo de paquetes software es muy importante para el alumno, puesto que este tipo de herramientas son básicas para desarrollar entornos de realidad acústica virtual donde se puedan realizar auralizaciones que permitan obtener una representación sonora del espacio simulado. De igual forma, este tipo de herramientas softwares son ampliamente utilizadas en la sonorización de eventos de gran envergadura. La tercera práctica incluye la medida del aislamiento a ruido aéreo entre locales como una forma de cuestionar la mayoría de requisitos que impone la normativa sobre este tipo de medidas, especialmente en lo referente a la medida de tiempo de reverberación y el muestreo del campo sonoro

    Combinations of adaptive filters

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    Adaptive filters are at the core of many signal processing applications, ranging from acoustic noise supression to echo cancelation [1], array beamforming [2], channel equalization [3], to more recent sensor network applications in surveillance, target localization, and tracking. A trending approach in this direction is to recur to in-network distributed processing in which individual nodes implement adaptation rules and diffuse their estimation to the network [4], [5].The work of Jerónimo Arenas-García and Luis Azpicueta-Ruiz was partially supported by the Spanish Ministry of Economy and Competitiveness (under projects TEC2011-22480 and PRI-PIBIN-2011-1266. The work of Magno M.T. Silva was partially supported by CNPq under Grant 304275/2014-0 and by FAPESP under Grant 2012/24835-1. The work of Vítor H. Nascimento was partially supported by CNPq under grant 306268/2014-0 and FAPESP under grant 2014/04256-2. The work of Ali Sayed was supported in part by NSF grants CCF-1011918 and ECCS-1407712. We are grateful to the colleagues with whom we have shared discussions and coauthorship of papers along this research line, especially Prof. Aníbal R. Figueiras-Vidal

    Colección de prácticas de instrumentación acústica y control de ruido

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    Grado en Ingeniería de Sistemas Audiovisuales. Asignatura: Instrumentación Acústica y Control de RuidoLa asignatura Instrumentación Acústica y Control de Ruido está encuadrada en el cuarto curso del Grado de Ingeniería de Sistemas Audiovisuales en la Universidad Carlos III de Madrid. Esta materia trata de formar a los estudiantes en lo que se refiere a cómo se diseña la instrumentación utilizada en la mayoría de las medidas acústicas, y a cómo hay que proceder en la realización de las medidas más habituales en acústica ambiental y de la edificación. El objetivo de esta asignatura es formar profesionales cualificados que puedan ejercer como técnicos especialistas y directores técnicos en laboratorios de acústica. Se definen 6 prácticas de laboratorio con el objetivo de que el alumno adquiera los conocimientos necesarios

    As light as your footsteps: design and evaluation of a portable device for changing body perception through a sound illusion

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    People’s body perception is highly malleable. Recent works have demonstrated that the dynamic modification of footstep sounds can lead people to perceive their body as thinner/lighter, walk more dynamically and feel happier, potentially supporting health. Previous studies modified the spectra of footstep sounds through a stereo 9-band analog graphic equalizer. While this system had minimal latency, it was not optimal as a wearable device, considering its weight (near 2 kg) and necessity of an electric outlet, which limited its applicability to real-world scenarios. Consequently, several substitute solutions were tested to improve portability, lightness and freedom of movement. For some, a non-satisfactory attempt was made to replicate the spectra of the original system. Therefore, it was hypothesized that a standalone digital microcomputer could increase portability and replicate the spectra. A novel device, using Bela.io and SuperCollider programming language, was tested, in which the spectral behavior of the original equalizer was replicated using cascaded biquad IIR filters. Objective and subjective experimental results suggest that, subject to the original system, we have successfully reduced weight and increased portability while keeping latency and spectral difference negligible. We foresee this novel system as a portable robust solution to induce illusory changes in body perception.This project has received funding from the European Research Council (ERC) under the European Union’s Horizon 2020 research and innovation programme (grant agreement No 101002711)

    Does the method matter? A review of the main testing methods for the subjective evaluation of room acoustics through listening tests.

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    Listening tests are experimental procedures in which a group of participants are asked to express their opinion about certain questions regarding some stimuli that they have been urged to listen to. The listening tests have proved their effectiveness as methods for the subjective assessment of sound perception in different fields of acoustics, and particularly in room acoustics and sound insulation. The way in which the listening test is conducted, how the participant's response is expressed and how the stimuli are presented to the participants define what it is known as the testing/query method. There is not yet a common methodological framework that defines which testing method is the best for each purpose and under what conditions the experiments are to be carried out. The purpose of this communication is to review the most commonly used testing methods in the subjective assessment of sound and to present the advantages and disadvantages of each of them when they are used for the evaluation of the subjective perception of a sample of population

    Does the method matter? A review of the main testing methods for the subjective evaluation of room acoustics through listening tests.

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    Listening tests are experimental procedures in which a group of participants are asked to express their opinion about certain questions regarding some stimuli that they have been urged to listen to. The listening tests have proved their effectiveness as methods for the subjective assessment of sound perception in different fields of acoustics, and particularly in room acoustics and sound insulation. The way in which the listening test is conducted, how the participant's response is expressed and how the stimuli are presented to the participants define what it is known as the testing/query method. There is not yet a common methodological framework that defines which testing method is the best for each purpose and under what conditions the experiments are to be carried out. The purpose of this communication is to review the most commonly used testing methods in the subjective assessment of sound and to present the advantages and disadvantages of each of them when they are used for the evaluation of the subjective perception of a sample of population
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