16 research outputs found

    Twenty-five years of sensor array and multichannel signal processing: a review of progress to date and potential research directions

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    In this article, a general introduction to the area of sensor array and multichannel signal processing is provided, including associated activities of the IEEE Signal Processing Society (SPS) Sensor Array and Multichannel (SAM) Technical Committee (TC). The main technological advances in five SAM subareas made in the past 25 years are then presented in detail, including beamforming, direction-of-arrival (DOA) estimation, sensor location optimization, target/source localization based on sensor arrays, and multiple-input multiple-output (MIMO) arrays. Six recent developments are also provided at the end to indicate possible promising directions for future SAM research, which are graph signal processing (GSP) for sensor networks; tensor-based array signal processing, quaternion-valued array signal processing, 1-bit and noncoherent sensor array signal processing, machine learning and artificial intelligence (AI) for sensor arrays; and array signal processing for next-generation communication systems

    Discrete interferences optimum beamformer in correlated signal and interfering noise

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    This paper introduces a significant special situation where the noise is a collection of D-plane interference signals and the correlated noise of D+1 is less than the number of array components. An optimal beamforming processor based on the minimum variance distortionless response (MVDR) generates and combines appropriate statistics for the D+1 model. Instead of the original space of the N-dimensional problem, the interference signal subspace is reduced to D+1. Typical antenna arrays in many modern communication networks absorb waves generated from multiple point sources. An analytical formula was derived to improve the signal to interference and noise ratio (SINR) obtained from the steering errors of the two beamformers. The proposed MVDR processor-based beamforming does not enforce general constraints. Therefore, it can also be used in systems where the steering vector is compromised by gain. Simulation results show that the output of the proposed beamformer based on the MVDR processor is usually close to the ideal state within a wide range of signal-to-noise ratio and signal-to-interference ratio. The MVDR processor-based beamformer has been experimentally evaluated. The proposed processor-based MVDR system significantly improves performance for large interference white noise ratio (INR) in the sidelobe region and provide an appropriate beam pattern

    Array Signal Processing Based on Traditional and Sparse Arrays

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    Array signal processing is based on using an array of sensors to receive the impinging signals. The received data is either spatially filtered to focus the signals from a desired direction or it may be used for estimating a parameter of source signal like direction of arrival (DOA), polarization and source power. Spatial filtering also known as beamforming and DOA estimation are integral parts of array signal processing and this thesis is aimed at solving some key probems related to these two areas. Wideband beamforming holds numerous applications in the bandwidth hungry data traffic of present day world. Several techniques exist to design fixed wideband beamformers based on traditional arrays like uniform linear array (ULA). Among these techniques, least squares based eigenfilter method is a key technique which has been used extensively in filter and wideband beamformer design. The first contribution of this thesis comes in the form of critically analyzing the standard eigenfilter method where a serious flaw in the design formulation is highlighted which generates inconsistent design performance, and an additional constraint is added to stabilize the achieved design. Simulation results show the validity and significance of the proposed method. Traditional arrays based on ULAs have limited applications in array signal processing due to the large number of sensors required and this problem has been addressed by the application of sparse arrays. Sparse arrays have been exploited from the perspective of their difference co-array structures which provide significantly higher number of degrees of freedoms (DOFs) compared to ULAs for the same number of sensors. These DOFs (consecutive and unique lags) are utilized in the application of DOA estimation with the help of difference co-array based DOA estimators. Several types of sparse arrays include minimum redundancy array (MRA), minimum hole array (MHA), nested array, prototype coprime array, conventional coprime array, coprime array with compressed interelement spacing (CACIS), coprime array with displaced subarrays (CADiS) and super nested array. As a second contribution of this thesis, a new sparse array termed thinned coprime array (TCA) is proposed which holds all the properties of a conventional coprime array but with \ceil*{\frac{M}{2}} fewer sensors where MM is the number of sensors of a subarray in the conventional structure. TCA possesses improved level of sparsity and is robust against mutual coupling compared to other sparse arrays. In addition, TCA holds higher number of DOFs utilizable for DOA estimation using variety of methods. TCA also shows lower estimation error compared to super nested arrays and MRA with increasing array size. Although TCA holds numerous desirable features, the number of unique lags offered by TCA are close to the sparsest CADiS and nested array and significantly lower than MRA which limits the estimation error performance offered by TCA through (compressive sensing) CS-based methods. In this direction, the structure of TCA is studied to explore the possibility of an array which can provide significantly higher number of unique lags with improved sparsity for a given number of sensors. The result of this investigation is the third contribution of this thesis in the form of a new sparse array, displaced thinned coprime array with additional sensor (DiTCAAS), which is based on a displaced version of TCA. The displacement of the subarrays generates an increase in the unique lags but the minimum spacing between the sensors becomes an integer multiple of half wavelength. To avoid spatial aliasing, an additional sensor is added at half wavelength from one of the sensors of the displaced subarray. The proposed placement of the additional sensor generates significantly higher number of unique lags for DiTCAAS, even more than the DOFs provided by MRA. Due to its improved sparsity and higher number of unique lags, DiTCAAS generates the lowest estimation error and robustness against heavy mutual coupling compared to super nested arrays, MRA, TCA and sparse CADiS with CS-based DOA estimation

    Sensor Array Processing with Manifold Uncertainty

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    <p>The spatial spectrum, also known as a field directionality map, is a description of the spatial distribution of energy in a wavefield. By sampling the wavefield at discrete locations in space, an estimate of the spatial spectrum can be derived using basic wave propagation models. The observable data space corresponding to physically realizable source locations for a given array configuration is referred to as the array manifold. In this thesis, array manifold ambiguities for linear arrays of omni-directional sensors in non-dispersive fields are considered. </p><p>First, the problem of underwater a hydrophone array towed behind a maneuvering platform is considered. The array consists of many hydrophones mounted to a flexible cable that is pulled behind a ship. The towed cable will bend or distort as the ship performs maneuvers. The motion of the cable through the turn can be used to resolve ambiguities that are inherent to nominally linear arrays. The first significant contribution is a method to estimate the spatial spectrum using a time-varying array shape in a dynamic field and broadband temporal data. Knowledge of the temporal spectral shape is shown to enhance detection performance. The field is approximated as a sum of uncorrelated planewaves located at uniform locations in angle, forming a gridded map on which a maximum likelihood estimate for broadband source power is derived. Uniform linear arrays also suffer from spatial aliasing when the inter-element spacing exceeds a half-wavelength. Broadband temporal knowledge is shown to significantly reduce aliasing and thus, in simulation, enhance target detection in interference dominated environments. </p><p>As an extension, the problem of towed array shape estimation is considered when the number and location of sources are unknown. A maximum likelihood estimate of the array shape using the field directionality map is derived. An acoustic-based array shape estimate that exploits the full 360^\circ field via field directionality mapping is the second significant contribution. Towed hydrophone arrays have heading sensors in order to estimate array shape, but these sensors can malfunction during sharp turns. An array shape model is described that allows the heading sensor data to be statistically fused with heading sensor. The third significant contribution is method to exploit dynamical motion models for sharp turns for a robust array shape estimate that combines acoustic and heading data. The proposed array shape model works well for both acoustic and heading data and is valid for arbitrary continuous array shapes.</p><p>Finally, the problem of array manifold ambiguities for static under-sampled linear arrays is considered. Under-sampled arrays are non-uniformly sampled with average spacing greater than a half-wavelength. While spatial aliasing only occurs in uniformly sampled arrays with spacing greater than a half-wavelength, under-sampled arrays have increased spatial resolution at the cost of high sidelobes compared to half-wavelength sampled arrays with the same number of sensors. Additionally, non-uniformly sampled arrays suffer from rank deficient array manifolds that cause traditional subspace based techniques to fail. A class of fully agumentable arrays, minimally redundant linear arrays, is considered where the received data statistics of a uniformly spaced array of the same length can be reconstructed in wide sense stationary fields at the cost of increased variance. The forth significant contribution is a reduced rank processing method for fully augmentable arrays to reduce the variance from augmentation with limited snapshots. Array gain for reduced rank adaptive processing with diagonal loading for snapshot deficient scenarios is analytically derived using asymptotic results from random matrix theory for a set ratio of sensors to snapshots. Additionally, the problem of near-field sources is considered and a method to reduce the variance from augmentation is proposed. In simulation, these methods result in significant average and median array gains with limited snapshots.</p>Dissertatio

    Design of large polyphase filters in the Quadratic Residue Number System

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    Temperature aware power optimization for multicore floating-point units

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    Effizientes binaurales Rendering von virtuellen akustischen Realitäten : technische und wahrnehmungsbezogene Konzepte

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    Binaural rendering aims to immerse the listener in a virtual acoustic scene, making it an essential method for spatial audio reproduction in virtual or augmented reality (VR/AR) applications. The growing interest and research in VR/AR solutions yielded many different methods for the binaural rendering of virtual acoustic realities, yet all of them share the fundamental idea that the auditory experience of any sound field can be reproduced by reconstructing its sound pressure at the listener's eardrums. This thesis addresses various state-of-the-art methods for 3 or 6 degrees of freedom (DoF) binaural rendering, technical approaches applied in the context of headphone-based virtual acoustic realities, and recent technical and psychoacoustic research questions in the field of binaural technology. The publications collected in this dissertation focus on technical or perceptual concepts and methods for efficient binaural rendering, which has become increasingly important in research and development due to the rising popularity of mobile consumer VR/AR devices and applications. The thesis is organized into five research topics: Head-Related Transfer Function Processing and Interpolation, Parametric Spatial Audio, Auditory Distance Perception of Nearby Sound Sources, Binaural Rendering of Spherical Microphone Array Data, and Voice Directivity. The results of the studies included in this dissertation extend the current state of research in the respective research topic, answer specific psychoacoustic research questions and thereby yield a better understanding of basic spatial hearing processes, and provide concepts, methods, and design parameters for the future implementation of technically and perceptually efficient binaural rendering.Binaurales Rendering zielt darauf ab, dass der Hörer in eine virtuelle akustische Szene eintaucht, und ist somit eine wesentliche Methode für die räumliche Audiowiedergabe in Anwendungen der virtuellen Realität (VR) oder der erweiterten Realität (AR – aus dem Englischen Augmented Reality). Das wachsende Interesse und die zunehmende Forschung an VR/AR-Lösungen führte zu vielen verschiedenen Methoden für das binaurale Rendering virtueller akustischer Realitäten, die jedoch alle die grundlegende Idee teilen, dass das Hörerlebnis eines beliebigen Schallfeldes durch die Rekonstruktion seines Schalldrucks am Trommelfell des Hörers reproduziert werden kann. Diese Arbeit befasst sich mit verschiedenen modernsten Methoden zur binauralen Wiedergabe mit 3 oder 6 Freiheitsgraden (DoF – aus dem Englischen Degree of Freedom), mit technischen Ansätzen, die im Kontext kopfhörerbasierter virtueller akustischer Realitäten angewandt werden, und mit aktuellen technischen und psychoakustischen Forschungsfragen auf dem Gebiet der Binauraltechnik. Die in dieser Dissertation gesammelten Publikationen befassen sich mit technischen oder wahrnehmungsbezogenen Konzepten und Methoden für effizientes binaurales Rendering, was in der Forschung und Entwicklung aufgrund der zunehmenden Beliebtheit von mobilen Verbraucher-VR/AR-Geräten und -Anwendungen zunehmend an Relevanz gewonnen hat. Die Arbeit ist in fünf Forschungsthemen gegliedert: Verarbeitung und Interpolation von Außenohrübertragungsfunktionen, parametrisches räumliches Audio, auditive Entfernungswahrnehmung ohrnaher Schallquellen, binaurales Rendering von sphärischen Mikrofonarraydaten und Richtcharakteristik der Stimme. Die Ergebnisse der in dieser Dissertation enthaltenen Studien erweitern den aktuellen Forschungsstand im jeweiligen Forschungsfeld, beantworten spezifische psychoakustische Forschungsfragen und führen damit zu einem besseren Verständnis grundlegender räumlicher Hörprozesse, und liefern Konzepte, Methoden und Gestaltungsparameter für die zukünftige Umsetzung eines technisch und wahrnehmungsbezogen effizienten binauralen Renderings.BMBF, 03FH014IX5, Natürliche raumbezogene Darbietung selbsterzeugter Schallereignisse in virtuellen auditiven Umgebungen (NarDasS

    Semi-coprime microphone arrays for estimating direction of arrival of speech sources

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    © 2019 IEEE. This paper evaluates the performance of semi-coprime microphone arrays (SCPMAs) for speech source direction of arrival (DOA) estimation based on the steered response power - phase transform (SRP-PHAT) algorithm. The SCPMA is an extension of the coprime microphone array (CPMA), which combines the outputs of three sub-arrays to reduce the impact of spatial aliasing and achieves performance comparable to that obtained from arrays using much larger numbers of microphones. The proposed approach considers two different processors to calculate the outputs from the sub-arrays and adapts the SRP-PHAT approach to these arrays. Simulations are conducted under anechoic and reverberant scenarios in a noisy room. Beam pattern and array gain results indicate that the SCPMA works better than the conventional CPMA at reducing the peak side lobe (PSL) level and total side lobe area while increasing the capability of amplifying the desired target signal and restraining noise from all other directions for typical frequencies of speeches. DOA Estimation results also show that the SCPMA achieves accurate DOA estimates in anechoic and low reverberant conditions, which is comparable to the equivalent full ULA, while the large side lobes in the beam pattern of the SCPMA lead to less accurate results in the highly reverberant environment
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