A miniature, lowpower , intelligent sensor node for persistent acoustic surveillance

ABSTRACT The desire for persistent, long term surveillance and covertness places severe constraints on the power consumption of a sensor node. To achieve the desired endurance while minimizing the size of the node, it is imperative to use application-specific integrated circuits (ASICs) that deliver the required performance with maximal power efficiency while minimizing the amount of communication bandwidth needed. This paper reviews our ongoing effort to integrate several micropower devices for low-power wake-up detection, blind source separation and localization and pattern classification, and demonstrate the utility of the system in relevant surveillance applications. The capabilities of each module are presented in detail along with performance statistics measured during recent experiments

Single- and multi-microphone speech dereverberation using spectral enhancement

In speech communication systems, such as voice-controlled systems, hands-free mobile telephones, and hearing aids, the received microphone signals are degraded by room reverberation, background noise, and other interferences. This signal degradation may lead to total unintelligibility of the speech and decreases the performance of automatic speech recognition systems. In the context of this work reverberation is the process of multi-path propagation of an acoustic sound from its source to one or more microphones. The received microphone signal generally consists of a direct sound, reflections that arrive shortly after the direct sound (commonly called early reverberation), and reflections that arrive after the early reverberation (commonly called late reverberation). Reverberant speech can be described as sounding distant with noticeable echo and colouration. These detrimental perceptual effects are primarily caused by late reverberation, and generally increase with increasing distance between the source and microphone. Conversely, early reverberations tend to improve the intelligibility of speech. In combination with the direct sound it is sometimes referred to as the early speech component. Reduction of the detrimental effects of reflections is evidently of considerable practical importance, and is the focus of this dissertation. More specifically the dissertation deals with dereverberation techniques, i.e., signal processing techniques to reduce the detrimental effects of reflections. In the dissertation, novel single- and multimicrophone speech dereverberation algorithms are developed that aim at the suppression of late reverberation, i.e., at estimation of the early speech component. This is done via so-called spectral enhancement techniques that require a specific measure of the late reverberant signal. This measure, called spectral variance, can be estimated directly from the received (possibly noisy) reverberant signal(s) using a statistical reverberation model and a limited amount of a priori knowledge about the acoustic channel(s) between the source and the microphone(s). In our work an existing single-channel statistical reverberation model serves as a starting point. The model is characterized by one parameter that depends on the acoustic characteristics of the environment. We show that the spectral variance estimator that is based on this model, can only be used when the source-microphone distance is larger than the so-called critical distance. This is, crudely speaking, the distance where the direct sound power is equal to the total reflective power. A generalization of the statistical reverberation model in which the direct sound is incorporated is developed. This model requires one additional parameter that is related to the ratio between the direct sound energy and the sound energy of all reflections. The generalized model is used to derive a novel spectral variance estimator. When the novel estimator is used for dereverberation rather than the existing estimator, and the source-microphone distance is smaller than the critical distance, the dereverberation performance is significantly increased. Single-microphone systems only exploit the temporal and spectral diversity of the received signal. Reverberation, of course, also induces spatial diversity. To additionally exploit this diversity, multiple microphones must be used, and their outputs must be combined by a suitable spatial processor such as the so-called delay and sum beamformer. It is not a priori evident whether spectral enhancement is best done before or after the spatial processor. For this reason we investigate both possibilities, as well as a merge of the spatial processor and the spectral enhancement technique. An advantage of the latter option is that the spectral variance estimator can be further improved. Our experiments show that the use of multiple microphones affords a significant improvement of the perceptual speech quality. The applicability of the theory developed in this dissertation is demonstrated using a hands-free communication system. Since hands-free systems are often used in a noisy and reverberant environment, the received microphone signal does not only contain the desired signal but also interferences such as room reverberation that is caused by the desired source, background noise, and a far-end echo signal that results from a sound that is produced by the loudspeaker. Usually an acoustic echo canceller is used to cancel the far-end echo. Additionally a post-processor is used to suppress background noise and residual echo, i.e., echo which could not be cancelled by the echo canceller. In this work a novel structure and post-processor for an acoustic echo canceller are developed. The post-processor suppresses late reverberation caused by the desired source, residual echo, and background noise. The late reverberation and late residual echo are estimated using the generalized statistical reverberation model. Experimental results convincingly demonstrate the benefits of the proposed system for suppressing late reverberation, residual echo and background noise. The proposed structure and post-processor have a low computational complexity, a highly modular structure, can be seamlessly integrated into existing hands-free communication systems, and affords a significant increase of the listening comfort and speech intelligibility

Kinematic State Estimation using Multiple DGPS/MEMS-IMU Sensors

Animals have evolved over billions of years and understanding these complex and intertwined systems have potential to advance the technology in the field of sports science, robotics and more. As such, a gait analysis using Motion Capture (MOCAP) technology is the subject of a number of research and development projects aimed at obtaining quantitative measurements. Existing MOCAP technology has limited the majority of studies to the analysis of the steady-state locomotion in a controlled (indoor) laboratory environment. MOCAP systems such as the optical, non-optical acoustic and non-optical magnetic MOCAP systems require predefined capture volume and controlled environmental conditions whilst the non-optical mechanical MOCAP system impedes the motion of the subject. Although the non-optical inertial MOCAP system allows MOCAP in an outdoor environment, it suffers from measurement noise and drift and lacks global trajectory information. The accuracy of these MOCAP systems are known to decrease during the tracking of the transient locomotion. Quantifying the manoeuvrability of animals in their natural habitat to answer the question “Why are animals so manoeuvrable?” remains a challenge. This research aims to develop an outdoor MOCAP system that will allow tracking of the steady-state as well as the transient locomotion of an animal in its natural habitat outside a controlled laboratory condition. A number of researchers have developed novel MOCAP systems with the same aim of creating an outdoor MOCAP system that is aimed at tracking the motion outside a controlled laboratory (indoor) environment with unlimited capture volume. These novel MOCAP systems are either not validated against the commercial MOCAP systems or do not have comparable sub-millimetre accuracy as the commercial MOCAP systems. The developed DGPS/MEMS-IMU multi-receiver fusion MOCAP system was assessed to have global trajectory accuracy of _0:0394m, relative limb position accuracy of _0:006497m. To conclude the research, several recommendations are made to improve the developed MOCAP system and to prepare for a field-testing with a wild animal from a family of a terrestrial megafauna

A Heterogeneous System Architecture for Low-Power Wireless Sensor Nodes in Compute-Intensive Distributed Applications

Wireless Sensor Networks (WSNs) combine embedded sensing and processing capabilities with a wireless communication infrastructure, thus supporting distributed monitoring applications. WSNs have been investigated for more than three decades, and recent social and industrial developments such as home automation, or the Internet of Things, have increased the commercial relevance of this key technology. The communication bandwidth of the sensor nodes is limited by the transportation media and the restricted energy budget of the nodes. To still keep up with the ever increasing sensor count and sampling rates, the basic data acquisition and collection capabilities of WSNs have been extended with decentralized smart feature extraction and data aggregation algorithms. Energy-efficient processing elements are thus required to meet the ever-growing compute demands of the WSN motes within the available energy budget. The Hardware-Accelerated Low Power Mote (HaLoMote) is proposed and evaluated in this thesis to address the requirements of compute-intensive WSN applications. It is a heterogeneous system architecture, that combines a Field Programmable Gate Array (FPGA) for hardware-accelerated data aggregation with an IEEE 802.15.4 based Radio Frequency System-on-Chip for the network management and the top-level control of the applications. To properly support Dynamic Power Management (DPM) on the HaLoMote, a Microsemi IGLOO FPGA with a non-volatile configuration storage was chosen for a prototype implementation, called Hardware-Accelerated Low Energy Wireless Embedded Sensor Node (HaLOEWEn). As for every multi-processor architecture, the inter-processor communication and coordination strongly influences the efficiency of the HaLoMote. Therefore, a generic communication framework is proposed in this thesis. It is tightly coupled with the DPM strategy of the HaLoMote, that supports fast transitions between active and idle modes. Low-power sleep periods can thus be scheduled within every sampling cycle, even for sampling rates of hundreds of hertz. In addition to the development of the heterogeneous system architecture, this thesis focuses on the energy consumption trade-off between wireless data transmission and in-sensor data aggregation. The HaLOEWEn is compared with typical software processors in terms of runtime and energy efficiency in the context of three monitoring applications. The building blocks of these applications comprise hardware-accelerated digital signal processing primitives, lossless data compression, a precise wireless time synchronization protocol, and a transceiver scheduling for contention free information flooding from multiple sources to all network nodes. Most of these concepts are applicable to similar distributed monitoring applications with in-sensor data aggregation. A Structural Health Monitoring (SHM) application is used for the system level evaluation of the HaLoMote concept. The Random Decrement Technique (RDT) is a particular SHM data aggregation algorithm, which determines the free-decay response of the monitored structure for subsequent modal identification. The hardware-accelerated RDT executed on a HaLOEWEn mote requires only 43 % of the energy that a recent ARM Cortex-M based microcontroller consumes for this algorithm. The functionality of the overall WSN-based SHM system is shown with a laboratory-scale demonstrator. Compared to reference data acquired by a wire-bound laboratory measurement system, the HaLOEWEn network can capture the structural information relevant for the SHM application with less than 1 % deviation

Towards a Common Software/Hardware Methodology for Future Advanced Driver Assistance Systems

The European research project DESERVE (DEvelopment platform for Safe and Efficient dRiVE, 2012-2015) had the aim of designing and developing a platform tool to cope with the continuously increasing complexity and the simultaneous need to reduce cost for future embedded Advanced Driver Assistance Systems (ADAS). For this purpose, the DESERVE platform profits from cross-domain software reuse, standardization of automotive software component interfaces, and easy but safety-compliant integration of heterogeneous modules. This enables the development of a new generation of ADAS applications, which challengingly combine different functions, sensors, actuators, hardware platforms, and Human Machine Interfaces (HMI). This book presents the different results of the DESERVE project concerning the ADAS development platform, test case functions, and validation and evaluation of different approaches. The reader is invited to substantiate the content of this book with the deliverables published during the DESERVE project. Technical topics discussed in this book include:Modern ADAS development platforms;Design space exploration;Driving modelling;Video-based and Radar-based ADAS functions;HMI for ADAS;Vehicle-hardware-in-the-loop validation system

Speech enhancement algorithms for audiological applications

Texto en inglés y resumen en inglés y españolPremio Extraordinario de Doctorado de la UAH en el año académico 2013-2014La mejora de la calidad de la voz es un problema que, aunque ha sido abordado durante muchos años, aún sigue abierto. El creciente auge de aplicaciones tales como los sistemas manos libres o de reconocimiento de voz automático y las cada vez mayores exigencias de las personas con pérdidas auditivas han dado un impulso definitivo a este área de investigación. Esta tesis doctoral se centra en la mejora de la calidad de la voz en aplicaciones audiológicas. La mayoría del trabajo de investigación desarrollado en esta tesis está dirigido a la mejora de la inteligibilidad de la voz en audífonos digitales, teniendo en cuenta las limitaciones de este tipo de dispositivos. La combinación de técnicas de separación de fuentes y filtrado espacial con técnicas de aprendizaje automático y computación evolutiva ha originado novedosos e interesantes algoritmos que son incluidos en esta tesis. La tesis esta dividida en dos grandes bloques. El primer bloque contiene un estudio preliminar del problema y una exhaustiva revisión del estudio del arte sobre algoritmos de mejora de la calidad de la voz, que sirve para definir los objetivos de esta tesis. El segundo bloque contiene la descripción del trabajo de investigación realizado para cumplir los objetivos de la tesis, así como los experimentos y resultados obtenidos. En primer lugar, el problema de mejora de la calidad de la voz es descrito formalmente en el dominio tiempo-frecuencia. Los principales requerimientos y restricciones de los audífonos digitales son definidas. Tras describir el problema, una amplia revisión del estudio del arte ha sido elaborada. La revisión incluye algoritmos de mejora de la calidad de la voz mono-canal y multi-canal, considerando técnicas de reducción de ruido y técnicas de separación de fuentes. Además, la aplicación de estos algoritmos en audífonos digitales es evaluada. El primer problema abordado en la tesis es la separación de fuentes sonoras en mezclas infra-determinadas en el dominio tiempo-frecuencia, sin considerar ningún tipo de restricción computacional. El rendimiento del famoso algoritmo DUET, que consigue separar fuentes de voz con solo dos mezclas, ha sido evaluado en diversos escenarios, incluyendo mezclas lineales y binaurales no reverberantes, mezclas reverberantes, y mezclas de voz con otro tipo de fuentes tales como ruido y música. El estudio revela la falta de robustez del algoritmo DUET, cuyo rendimiento se ve seriamente disminuido en mezclas reverberantes, mezclas binaurales, y mezclas de voz con música y ruido. Con el objetivo de mejorar el rendimiento en estos casos, se presenta un novedoso algoritmo de separación de fuentes que combina la técnica de clustering mean shift con la base del algoritmo DUET. La etapa de clustering del algoritmo DUET, que esta basada en un histograma ponderado, es reemplazada por una modificación del algoritmo mean shift, introduciendo el uso de un kernel Gaussiano ponderado. El análisis de los resultados obtenidos muestran una clara mejora obtenida por el algoritmo propuesto en relación con el algoritmo DUET original y una modificación que usa k-means. Además, el algoritmo propuesto ha sido extendido para usar un array de micrófonos de cualquier tamaño y geometría. A continuación se ha abordado el problema de la enumeración de fuentes de voz, que esta relacionado con el problema de separación de fuentes. Se ha propuesto un novedoso algoritmo basado en un criterio de teoría de la información y en la estimación de los retardos relativos causados por las fuentes entre un par de micrófonos. El algoritmo ha obtenido excelente resultados y muestra robustez en la enumeración de mezclas no reverberantes de hasta 5 fuentes de voz. Además se demuestra la potencia del algoritmo para la enumeración de fuentes en mezclas reverberantes. El resto de la tesis esta centrada en audífonos digitales. El primer problema tratado es el de la mejora de la inteligibilidad de la voz en audífonos monoaurales. En primer lugar, se realiza un estudio de los recursos computacionales disponibles en audífonos digitales de ultima generación. Los resultados de este estudio se han utilizado para limitar el coste computacional de los algoritmos de mejora de la calidad de la voz para audífonos propuestos en esta tesis. Para resolver este primer problema se propone un algoritmo mono-canal de mejora de la calidad de la voz de bajo coste computacional. El objetivo es la estimación de una mascara tiempo-frecuencia continua para obtener el mayor parámetro PESQ de salida. El algoritmo combina una versión generalizada del estimador de mínimos cuadrados con un algoritmo de selección de características a medida, utilizando un novedoso conjunto de características. El algoritmo ha obtenido resultados excelentes incluso con baja relación señal a ruido. El siguiente problema abordado es el diseño de algoritmos de mejora de la calidad de la voz para audífonos binaurales comunicados de forma inalámbrica. Estos sistemas tienen un problema adicional, y es que la conexión inalámbrica aumenta el consumo de potencia. El objetivo en esta tesis es diseñar algoritmos de mejora de la calidad de la voz de bajo coste computacional que incrementen la eficiencia energética en audífonos binaurales comunicados de forma inalámbrica. Se han propuesto dos soluciones. La primera es un algoritmo de extremado bajo coste computacional que maximiza el parámetro WDO y esta basado en la estimación de una mascara binaria mediante un discriminante cuadrático que utiliza los valores ILD e ITD de cada punto tiempo-frecuencia para clasificarlo entre voz o ruido. El segundo algoritmo propuesto, también de bajo coste, utiliza además la información de puntos tiempo-frecuencia vecinos para estimar la IBM mediante una versión generalizada del LS-LDA. Además, se propone utilizar un MSE ponderado para estimar la IBM y maximizar el parámetro WDO al mismo tiempo. En ambos algoritmos se propone un esquema de transmisión eficiente energéticamente, que se basa en cuantificar los valores de amplitud y fase de cada banda de frecuencia con un numero distinto de bits. La distribución de bits entre frecuencias se optimiza mediante técnicas de computación evolutivas. El ultimo trabajo incluido en esta tesis trata del diseño de filtros espaciales para audífonos personalizados a una persona determinada. Los coeficientes del filtro pueden adaptarse a una persona siempre que se conozca su HRTF. Desafortunadamente, esta información no esta disponible cuando un paciente visita el audiólogo, lo que causa perdidas de ganancia y distorsiones. Con este problema en mente, se han propuesto tres métodos para diseñar filtros espaciales que maximicen la ganancia y minimicen las distorsiones medias para un conjunto de HRTFs de diseño

Locating and extracting acoustic and neural signals

This dissertation presents innovate methodologies for locating, extracting, and separating multiple incoherent sound sources in three-dimensional (3D) space; and applications of the time reversal (TR) algorithm to pinpoint the hyper active neural activities inside the brain auditory structure that are correlated to the tinnitus pathology. Specifically, an acoustic modeling based method is developed for locating arbitrary and incoherent sound sources in 3D space in real time by using a minimal number of microphones, and the Point Source Separation (PSS) method is developed for extracting target signals from directly measured mixed signals. Combining these two approaches leads to a novel technology known as Blind Sources Localization and Separation (BSLS) that enables one to locate multiple incoherent sound signals in 3D space and separate original individual sources simultaneously, based on the directly measured mixed signals. These technologies have been validated through numerical simulations and experiments conducted in various non-ideal environments where there are non-negligible, unspecified sound reflections and reverberation as well as interferences from random background noise. Another innovation presented in this dissertation is concerned with applications of the TR algorithm to pinpoint the exact locations of hyper-active neurons in the brain auditory structure that are directly correlated to the tinnitus perception. Benchmark tests conducted on normal rats have confirmed the localization results provided by the TR algorithm. Results demonstrate that the spatial resolution of this source localization can be as high as the micrometer level. This high precision localization may lead to a paradigm shift in tinnitus diagnosis, which may in turn produce a more cost-effective treatment for tinnitus than any of the existing ones

NASA patent abstracts bibliography: A continuing bibliography. Section 1: Abstracts (supplement 40)

Abstracts are provided for 181 patents and patent applications entered into the NASA scientific and technical information system during the period July 1991 through December 1991. Each entry consists of a citation, an abstract, and in most cases, a key illustration selected from the patent or patent application