Spatial Active noise control (ANC) systems are proposed to minimize the noise over a spatial region of interest around people’s heads by generating an anti-noise field with multiple microphones and loudspeakers. Recently, a realistic microphone geometry was designed to allow the system to monitor the residual noise field without restricting people’s head movement. However, this design, which utilizes the remote microphone technique using an observation filter (OF) for noise field modelling, has a drawback that the system is not robust against variations in the noise source location. In this paper, we address this drawback and present an improved spatial ANC system that overcomes this limitation. The proposed system enhances its adaptability by selectively employing the most suitable OF from a set of pre-modelled discriminator filter (DF), tailored to the acoustic environment of the system. We demonstrate that the proposed method maintains noise reduction performance over the region of interest with a dynamic noise source with changing position and signal through simulation and physical measurements.

Active Noise Control (ANC) over large regions of interest (ROI) traditionally requires numerous evenly distributed error microphones, which is often impractical and obstructive for human occupants. In this paper, we proposed a wave domain adaptive ANC algorithm using the joint information from multiple error spherical microphone arrays (SMAs) on the boundary of the ROI. By mapping the SMA recordings to the virtual sound field and introducing ℓ1norm on the virtual plane wave weights, the proposed method can achieve good noise reduction over the entire larger ROI, especially when the number of noise sources is finite and sparse. Our finding presents a significant advancement in spatial ANC, paving the way for more effective and unobtrusive solutions in a range of environments.

Spatial audio reproduction and translation for virtual, augmented, and extended reality applications require an efficient representation of the recorded sound fields. In this paper, we investigate the possible sparse representations of the sound field recorded by multiple microphones in reverberant environments. We combine the Complex Orthogonal Matching Pursuit (COMP) algorithm with the concept of distributed virtual sound sources to propose a sparse sound field representation. The technique uses recordings from a grid of microphones and transforms them into a sparse representation featuring a selected set of active virtual sources. Using simulation, we evaluate the proposed COMP approach with LASSO and IRLS methods in a reverberant room of regular size with a ceiling-mounted microphone array.

Conventional multiple-point active noise control (ANC) systems require placing error microphones within the region of interest (ROI), inconveniencing users. This paper designs a feasible monitoring microphone arrangement placed outside the ROI, providing a user with more freedom of movement. The soundfield within the ROI is interpolated from the microphone signals using a physics-informed neural network (PINN). PINN exploits the acoustic wave equation to assist soundfield interpolation under a limited number of monitoring microphones, and demonstrates better interpolation performance than the spherical harmonics method in simulations. An ANC system is designed to take advantage of the interpolated signal to reduce noise signal within the ROI. The PINN-assisted ANC system reduces noise more than that of the multiple-point ANC system in simulations.

Acoustic velocity vectors (AVVs) are related to the human's perception of sound at low frequencies and are widely used in Ambisonics. This letter proposes a spatial sound field reproduction algorithm called velocity matching, which reproduces the AVVs in the spherical listening region by matching the AVVs' spherical harmonic coefficients. Using the sound field translation formula, the spherical harmonic coefficients of the AVVs are derived from the spherical harmonic coefficients of the pressure, which can be measured by a higher-order microphone array. Unlike algorithms that only control the AVVs at discrete sweet spots, the proposed velocity matching algorithm manipulates the AVVs in the whole spherical listening region and allows the listener to move beyond the sweet spots. Simulations show the proposed velocity matching algorithm accurately reproduces the AVVs in the spherical listening region and requires fewer number of loudspeakers than pressure matching algorithm.

Acoustic imaging is powerful in collecting spatial information of acoustic sources into a visual representation. In this paper, we focus on the analysis of the exterior acoustic field captured by a circular array of microphones. With a proper parametrization based on angles, we map the directions of arrival of sources as a function of the microphone locations, thus obtaining an acoustic image called “angular space”. Therefore, we introduce a linear transform to enable analysis and synthesis operations for mapping the microphone pressures onto the angular space using local space-time Fourier analysis. We prove the ability of this representation to combine global information coming from multiple arrays in a single acoustic image that can be processed and manipulated. Examples of source localization applications in simulated and measured scenarios show the effectiveness of the proposed method obtaining results comparable with state-of-the-art methods.

Large-scale Room Impulse Response (RIR) measurements are required to accurately determine a room’s acoustic response to different source-listener configurations. RIR reconstruction methods are often used to reduce these measurement costs. Prior knowledge of room acoustic parameters can ensure reliable and robust RIR reconstruction. This paper proposes a method to reconstruct RIRs based on reflection source locations and time-frequency-direction-dependent reflection magnitude response estimated from a single spherical microphone array measurement. These input parameters are learned by applying the eigenbeam spatial correlation method and von Mises–Fisher (vMF)-based directivity modeling. According to the performance evaluation, the composition of the learned features in the RIR reconstruction formulation successfully preserves the objective characteristics of the real room.

The development of spherical sector harmonics benefits the sound- field decomposition and analysis over a spherical sector region. However, research on the soundfield radial extrapolation from one spherical sector region to another concentric sector region with a different radius is still insufficient. This paper presents a radial extrapolation method for spherical sector soundfields. Based on the extrapolation error and error sensitivity ratio, the performance and robustness of the proposed method is analyzed with different setups. Results show that the proposed method can achieve good performance, especially with a dual array sensor placement, a large angle range of the measurement region, and a small wave number.

In literature, sound source localization under the far- and near-field scenarios are mostly addressed as independent tasks using different approaches. This causes a tedious task to detect the type of sound-field, whereas in practice there may not be a clear boundary between the far- and near-field soundfield. In contrast, this paper proposes a multi-channel feature denoted generalized relative harmonic coefficients (generalized RHC) in the spherical harmonics domain, which can equally localize both far- and near-field sound source without requiring any adjustments. We derive the analytical expression of this feature and summarize its unique properties, which facilitate two single-source directional-of-arrival estimators: (i) using a full grid search over the directional space; and (ii) a closed-form solution without any grid search. Experimental study in realistic noisy and reverberant environments under both near-field and far-field conditions validates the efficacy of the proposed algorithm.

Blind source separation aims to recover a series of unmixed signals from only a mixture recording of source signals. For the mixture of speech signals where sources have unique spatial positions, the relative transfer function has been found useful as it contains the spatial information of each source regardless of its signal. In this paper, we transfer a previous relative transfer function based separation method into the wave domain by utilizing a higher-order microphone for the mixture recording. Consequently, the higher-order microphone enables the processing with relative harmonic coefficients, which provides better independency between different coefficient components than the conventional relative transfer functions in each time frame. We demonstrate that the proposed method using relative harmonic coefficients outperforms the relative transfer function based method in highly reverberant rooms through simulation with both simulated and real measured room impulse response data.

Spatial active noise control (ANC) aims to reduce unwanted acoustic noise over a continuous spatial region by generating an anti-noise field with secondary loudspeakers. Conventionally, spatial ANC is achieved by using complex error microphone arrays such as grid or spherical geometry, which are impractical and obstruct the users to enter the quiet region. Recently, a spatial ANC system using circular arrays of microphones has been proposed such that the lower hemisphere of the region of interest is free of microphones for user’s head movement. In this structure, however, the error microphone located at the center of the region (the origin) blocks the head from moving up. In this paper, we optimize the aforementioned system in terms of the error microphone geometry. We remove the requirement to have a microphone at the origin by introducing a single-channel remote microphone technique to the spatial ANC system. Through real-world experiments, we demonstrate that the proposed method achieves noise reduction over space with a more user-friendly error microphone array design.

This paper presents the image source method for simulating the observed signals in the time-domain on the boundary of a spherical listening region. A wideband approach is used where all derivations are in the time-domain. The source emits a sequence of spherical wave fronts whose amplitudes could be related to the far-field directional impulse responses of a loudspeaker. Geometric methods are extensively used to model the observed signals. The spherical harmonic coefficients of the observed signals are also derived.

Fixed beamforming based on uniform linear microphone arrays often suffers from non-optimal performance for broadband signals. This paper addresses the issue by jointly optimizing the array geometry and spatial filters through a neural network based model. The model, composed of two feed forward neural networks, is optimized in an end-to-end manner. It satisfies the distortionless constraint in the look direction. Experimental results show that the proposed model outperforms the previous state-of-the-art fixed beamformer with overall better scores. Moreover, the proposed model can control the tradeoff between Directivity Factor (DF) and White Noise Gain (WNG) in a flexible way.

This paper presents the time-domain wideband spherical microphone array impulse response generator (TDW-SMIR generator), which is a time-domain wideband image source method (ISM) for generating the room impulse responses captured by an open spherical microphone array. To incorporate loudspeaker directivity, the TDW-SMIR generator considers a source that emits a sequence of spherical wave fronts whose amplitudes are related to the loudspeaker directional impulse responses measured in the far-field. The TDW-SMIR generator uses geometric models to derive the time-domain signals recorded by the spherical microphone array. Comparisons are made with frequency-domain single band ISMs. Simulation results prove the results of the TDW-SMIR generator are similar to those of frequency-domain single band ISMs.

Spatial active noise control (ANC) aims to reduce unwanted acoustic noise over a continuous spatial region by generating an anti-noise field with secondary loudspeakers. Conventionally, spatial ANC is achieved by using complex error microphone arrays such as grid or spherical geometry, which are impractical and obstruct the users to enter the quiet region. Recently, the remote microphone technique has been introduced to spatial ANC systems without using error microphones inside the region of interest. However, this technique still requires an error microphone array during the tuning stage. In this paper, we further improve the remote microphone technique by introducing a spatial sound field recording method with a moving higher order microphone for the noise field recording (tuning stage) as well as secondary channel estimations (control stage). This eliminates the requirement for impractical microphone array geometries, typically required in existing spatial ANC solutions. The experimental data based simulation demonstrates the effectiveness of the proposed method on noise reduction over space with its feasible array design.

Direction-of-arrival (DOA) estimation for multiple simultaneous speakers in reverberant environments is still one of the challenging tasks in the audio signal processing field. A recent approach addresses this problem using a spherical harmonics domain feature namedrelative harmonic coefficients(RHC). Based on a bin-wise operation across the STFT (short-time Fourier transform) domain, this method detects the direct-path RHC in the first stage, followed by single source localization in the second stage. However, the method is computationally expensive as each STFT bin requires an exhaustive grid search over the two-dimensional (2-D) directional space. In this paper, we propose a significantly more computationally efficient alternative that decouples the azimuth and elevation 2-D search to two separate one-dimensional (1-D) search. The proposed multi-speaker localization algorithm comprises of two main steps, responsible for: (i) achieving a joint direct-path RHC detection and decoupled DOA estimation using 1-D search; and (ii) counting the number of speakers and estimating their DOAs based on the estimates from direct-path dominated STFT bins. Experiments using both simulated and real-life reverberant recordings confirm the significant computational complexity reduction while achieving competitive localization accuracy, compared to the baseline approaches. Although our proposed method performs in an unsupervised manner, it proves to be applicable even under unfavorable acoustic environments with a high reverberation level (e.g.,$T_{60}=1$second).

This paper presents a sound source localization method using an irregular microphone array embedded in a drone. Sound source localization is an integral function of drone audition systems which enables various applications of drones such as search and rescue missions. However, the audio recordings using the on-board microphones obscure the sound emitted by a source on the ground due to drone generated motor and propeller noise, thus leading to an extremely low signal-to-drone noise ratio (SdNR). In this paper, we propose a cross-correlation based direction of arrival (DOA) estimation technique using the time difference of arrival (TDOA) at different microphone pairs, with noise angular spectrum subtraction. Through the measured current-specific drone noise spectrum, noise suppression has been achieved from the multi-channel recordings. Experimental results show that the proposed method is capable of estimating the position in three-dimensional space for simultaneously active multiple sound sources on the ground at low SdNR conditions ($-30$dB), and localize two sound sources located at a certain azimuth angular separation with low prediction error comparable to the multiple signal classification (MUSIC) based algorithms and the generalized cross-correlation with phase transformation (GCC-PHAT) method. Due to its simplicity, applicability to any array geometry, and better robustness against drone noise, the proposed method increases the feasibility of localization under extreme SdNR levels.

Microphone array calibration is required to accurately capture the information in an audio source recording. Existing calibration methods require expensive hardware and setup procedures to compute filters for correcting microphone responses. Typically, such methods struggle to extend measurement accuracy to low frequencies. As a result, the error due to microphone gain mismatch propagates to all the modes in the spherical harmonic domain representation of a signal. Several existing algorithms use modal representation of sound and error propagation in modes fundamentally limits the performance of such algorithms. A method for reducing the error propagation in modes by correcting the mismatched microphone gains is proposed, where RAndom PerturbatIons for Diffuse-field (RAPID) is used to design filters for correcting the mismatch. Experimental results show that the directivity pattern of a calibrated spherical microphone array using RAPID provides up to 6dB improvement in the front back factor.

A spherical harmonics domain source feature called relative harmonic coefficients (RHC) has recently been applied to address the source direction-of-arrival (DOA) estimation problem. This paper presents a compact evaluation and comparison between two existing RHC based DOA estimators: (i) a method using a full grid search over the two-dimensional (2-D) directional space, (ii) a decoupled estimator which uses one-dimensional (1-D) search to separately localize the source's elevation and azimuth. We also propose a new estimator using a gradient descent search over the 2-D directional grid space. Extensive experiments in both simulated and real-life environments are conducted to examine and analyze the performance of all the underlying DOA estimators. Two objective metrics, including localization accuracy and algorithm complexity, are adopted for an evaluation and comparison between all estimators.

Sound intensity is a good predictor of human perception of sound location, which can be controlled to provide impressive direction perception to humans in soundfield reproduction systems, especially when the loudspeakers are non-uniformly distributed. However, the previous works in this field are all constrained to a single sweet spot/spatial zone. We address this challenge and propose a multizone reproduction method for 3D soundfield in a reverberant room based on intensity matching. We develop spatial sound intensity expressions in a reververant room using spherical harmonic decomposition, and build a cost function to optimize sound intensity within multiple spatial zones. Finally, simulation results showing the performance are presented.

Non-interactive and linear experienceslike cinema film offer high quality surround sound audio to enhance immersion, however, the perspective is usually fixed to the recording microphone position. With the rise of virtual reality, there is a demand for recording and recreating real-world experiences that allow users to move throughout the reproduction. Sound field translation achieves this by building an equivalent environment of virtual sources to recreate the recording spatially. However, the technique remains to restrict the maximum distance a user can translate away from the recording microphone's perspective due to the discrete sampling by commercial higher order microphones only being capable of recording an acoustic sweet-spot. In this paper, we propose a method for binaurally reproducing a microphone recording in a virtual application that allows the user to freely translate their body further beyond the recording position. The method incorporates a mixture of near-field and far-field sources in a sparsely expanded virtual environment to maintain a perceptually accurate reproduction. We perceptually validate the method through a Multiple Stimulus with Hidden Reference and Anchor (MUSHRA) experiment. Compared to the planewave benchmark, the proposed method offers both improved source localizability and robustness to spectral distortions at translated listening positions. A cross-examination with numerical simulations demonstrated that the sparse expansion relaxes the inherent sweet-spot constraint, leading to the improved localizability for sparse environments. Additionally, the proposed method is seen to better reproduce the intensity and binaural room impulse response spectra of near-field environments, further supporting the perceptual results.

Subspace approach of MUSIC (multiple signal classication) has become one of the most popular multi-source direction of arrival (DOA) estimations due to its easy implementation in practice. However, its localization accuracy is vulnerable to noise. This paper develops a novel MUSIC algorithm, more suitable in noisy environments, using the relative sound pressure measurements of a higher order microphone array. This proposed MUSIC approach is also decomposed into the spherical harmonics domain where a frequency smoothing technique is allowed to de-correlate the coherent source signals for improved localization accuracy. The proposed algorithm is also capable of estimating the number of active sound sources, which is pre-requisite knowledge for the traditional MUSIC approach. Extensive experimental results in diverse environments using both simulated and real recordings show advantages of the proposed algorithm over the traditional MUSIC method as well as another recently proposed multi-source localization approach.

This article proposes multiple circular arrays of vector sensors for analyzing the three dimensional sound field. By exploiting the fact that a finite number of spatial basis functions can represent the sound field within a region, the designed arrays allow the analysis of the sound field around the arrays sequentially by either a frequency-domain method or a time-domain method. The arrays are compact and thus are suitable for integrating with small-sized devices. The time-domain method is of low latency and computationally simple, and thus are suitable for providing the devices with spatial acoustic features in real-time. The effectiveness of the proposed arrays and the sound field analysis methods are confirmed by simulations, and compared with another array.

All spatial active noise control (ANC) systems, when deployed in typical room environments, have time-varying acoustic channels between the secondary sources and the error microphones. The conventional online secondary path modeling techniques, which introduces additive auxiliary random noise to estimate the secondary paths, become challenging especially in a multichannel setup. In this work, we propose to use higher-order variable-directivity sound sources as secondary sources for spatial ANC, in which both the interior residual noise field within the control region and exterior sound field due to secondary source radiation are jointly controlled. The aim of controlling the exterior sound field is to minimize room reverberation generated by the secondary sources so that the secondary paths in the proposed algorithm can be approximated as free-field propagation and thus can be pre-calibrated. The system is implemented in an adaptive manner to track noise variations. The results show that the proposed method can effectively cancel spatial noise field and control exterior sound field at an acceptable low level in time-varying room environments.

Detection of statistical dependence between random variables is an essential component in many machine learning algorithms. We propose a novel independence criterion for two random variables with linear-time complexity. We establish that our independence criterion is an upper bound of the Hirschfeld-Gebelein-Rényi maximum correlation coefficient between tested variables. A finite set of basis functions is employed to approximate the mapping functions that can achieve the maximal correlation. Using classic benchmark experiments based on independent component analysis, we demonstrate that our independence criterion performs comparably with the state-of-the-art quadratic-time kernel dependence measures like the Hilbert-Schmidt Independence Criterion, while being more efficient in computation. The experimental results also show that our independence criterion outperforms another contemporary linear-time kernel dependence measure, the Finite Set Independence Criterion. The potential application of our criterion in deep neural networks is validated experimentally.

Active noise control (ANC) over space is a well-researched topic where multi-microphone, multi-loudspeaker systems are designed to minimize the noise over a spatial region of interest. In this paper, we perform an initial study on the more complex problem of simultaneous noise control over multiple target regions using a single ANC system. In particular, we investigate the maximum active noise control performance over the multiple target regions, given a particular setup of secondary loudspeakers. The performance analysis is carried out using a wave-domain representation to best represent sound propagation over multiple regions in a given enclosure. Furthermore, given the global primary noise field and a fixed secondary source setup, a subspace method is exploited to evaluate the best system performance that could minimize the residual noise fields over multiple regions of interest. We provide experimental results to demonstrate the effectiveness of the proposed method.

We propose a straightforward and cost-effective method to perform diffuse soundfield measurements for calibrating the magnitude response of a microphone array. Typically, such calibration is performed in a diffuse soundfield created in reverberation chambers, an expensive and time-consuming process. A method is proposed for obtaining diffuse field measurements in untreated environments. First, a closed-form expression for the spatial correlation of a wideband signal in a diffuse field is derived. Next, we describe a practical procedure for obtaining the diffuse field response of a microphone array in the presence of a non-diffuse soundfield by the introduction of random perturbations in the microphone location. Experimental spatial correlation data obtained is compared with the theoretical model, confirming that it is possible to obtain diffuse field measurements in untreated environments with relatively few loudspeakers. A 30 second test signal played from 4-8 loudspeakers is shown to be sufficient in obtaining a diffuse field measurement using the proposed method. An Eigenmike®is then successfully calibrated at two different geographical locations.

This paper presents an unsupervised multi-source localization algorithm using a recently introduced feature called the relative harmonic coefficients. We derive a closed-form expression of the feature and briefly summarize its unique properties. We then exploit this feature to develop a single-source frame/bin detector which simplifies the challenging problem of multiple source localization into a single source localization problem. We show that the underlying method is suitable for localization using overlapped, disjoint as well as simultaneous multi-source recordings. Experimental results in both simulated and real-life reverberant environments confirm improved localization accuracy of the proposed method in comparison with the existing state-of-art approach.

Over recent years, spatial acoustic signal processing using higher order microphone arrays in the spherical harmonics domain has been a popular research topic. This paper uses a recently introduced source feature called the relative harmonic coefficients to develop an acoustic signal enhancement approach in noisy environments. This proposed method enables to extract the clean spherical harmonic coefficients from noisy higher order microphone recordings. Hence, this technique can be used as a pre-processing tool for noise-free measurements required by many spatial audio applications. We finally present a simulation study analyzing the performance of this approach in far field noisy environments.

This work presents a method that persuades acoustic reflections to be a favorable property for sound source localization. Whilst most real world spatial audio applications utilize prior knowledge of sound source position, estimating such positions in reverberant environments is still considered to be a difficult problem due to acoustic reflections. This article presents a novel MUSIC framework for multiple sound source localization (range, elevation, azimuth) in reverberant rooms by incorporating a recently proposed region-to-region room transfer model. The method is built upon the received signals of a higher order microphone and a spherical harmonic representation of the room transfer function. We demonstrate the method's general applicability and multiple source localization performance through a simulation study across an assortment of reverberant conditions. Additionally, we investigate robustness against various system modeling errors to gauge implementation viability. Finally, we prove the method in a practical experiment inside a real-world room with measured region-to-region transfer function parameters.

We propose a novel multi-source direction of arrival (DOA) estimation technique using a convolutional neural network algorithm which learns the modal coherence patterns of an incident soundfield through measured spherical harmonic coefficients. We train our model for individual time-frequency bins in the short-time Fourier transform spectrum by analyzing the unique snapshot of modal coherence for each desired direction. The proposed method is capable of estimating simultaneously active multiple sound sources on a 3D space using a single-source training scheme. This single-source training scheme reduces the training time and resource requirements as well as allows the reuse of the same trained model for different multi-source combinations. The method is evaluated against various simulated and practical noisy and reverberant environments with varying acoustic criteria and found to outperform the baseline methods in terms of DOA estimation accuracy. Furthermore, the proposed algorithm allows independent training of azimuth and elevation during a full DOA estimation over 3D space which significantly improves its training efficiency without affecting the overall estimation accuracy.

This article develops a semi-supervised algorithm to address the challenging multi-source localization problem in a noisy and reverberant environment, using a spherical harmonics domain source feature of the relative harmonic coefficients. We present a comprehensive research of this source feature, including (i) an illustration confirming its sole dependence on the source position, (ii) a feature estimator in the presence of noise, (iii) a feature selector exploiting its inherent directivity over space. Source features at varied spherical harmonic modes, representing unique characterization of the soundfield, are fused by the Multi-Mode Gaussian Process modeling. Based on the unifying model, we then formulate the mapping function revealing the underlying relationship between the source feature(s) and position(s) using a Bayesian inference approach. Another issue of the overlapped components is addressed by a pre-processing technique performing overlapped frame detection, which in turn reduces this challenging problem to a single source localization. It is highlighted that this data-driven method has a strong potential to be implemented in practice because only a limited number of labeled measurements is required. We evaluate this proposed algorithm using simulated recordings between multiple speakers in diverse environments, and extensive results confirm improved performance in comparison with the state-of-art methods. Additional assessments using real-life recordings further prove the effectiveness of the method, even at unfavorable circumstances with severe source overlapping.

Active noise control system has been actively researched over the past half century, and implemented to reduce noises in ducts, headsets, and inside several automobile models. However, active control of noise fields, and specifically broadband noise fields, in rooms is still a barely explored topic due to the difficult of obtaining the reference signals, the challenge of online secondary path estimation, and the causal control constraint. In this paper, an active noise control system is developed that can cancel outgoing broadband noise fields in rooms. The proposed system decomposes the noise field on a sphere surrounding the noise sources into spherical harmonic modes, exploiting their directionality to generate the reference signals and remove the need for online secondary path estimation. A time-domain sound field separation algorithm and a time-wave domain adaptive algorithm allow the proposed system to meet the causal control constraint. Simulation results demonstrate that the proposed system can cancel broadband noise field globally in a room without suffering from the secondary source feedback problem.

Active acoustic noise attenuation over a sizable space is a challenging problem in signal processing. The noise attenuation performance of feedforward active noise control (ANC) relies on the preciseness of a reference signal of a primary noise field. To capture the precise reference signal for controlling a sizable space, a large number of reference microphones are required, which reduces system viability. In this study, we exploit an efficient representation of the reference signal in spherical harmonic (SH) domain by utilizing the inherent sparseness of the noise field. The main contributions of this work are as follows. (1) A general reference microphone geometry can be used. The implementation difficulty in the array structure, which is recognized as the common issue of SH-domain signal processing, e.g., use of a fully surrounding spherical array, is reduced by using the fields translation based on the addition theorem. (2) The accuracy of low-frequency signal decomposition is improved. The low accuracy of low-frequency signal decomposition in compressive sensing (CS), which is commonly reported in the literature, is improved by applying signal representation in SH domain. (3) System robustness is increased. The robustness of the system is increased by considering a noise source spatial distribution of both the interior and exterior sound fields, which is not possible in the case of a general signal representation in SH domain. Experimental results indicate that the noise attenuation performance of our proposed method exceeds that of existing solutions. The flexibility of the array structure is also increased, which leads to a more feasible practical system setup.

Spatial active noise control (ANC) systems focus on minimizing unwanted acoustic noise over a continuous spatial region. Conventionally, spatial ANC systems are proposed using point based multi-channel systems and recently novel methods have been developed using spherical harmonic analysis of spatial sound fields. A major limitation for implementing the latter approach is the requirement of regularly distributed microphones and loudspeakers over spherical surfaces. In this paper, we relax the above constraint by constructing a system utilizing multiple circular microphone and loudspeaker arrays and designing a corresponding ANC algorithm. By simulation, we show that the proposed method can achieve comparable ANC performance to conventional spherical array methods. By experiment, we demonstrate the feasibility of implementing the multiple circular array structure and verify its effectiveness given practical constraints.

In literature, particle velocity has been introduced to improve performance of spatial sound field reproduction systems. However, all existing work requires to have accurate particle velocity measurements at all of the discrete control points, which is difficult to obtain in real-world applications. In this work, we formulate continuous particle velocity expressions over space as a function of pressure coefficients in the modal domain that can be easily extracted by using a higher order microphone. The sound field within a target region is controlled by a weighted cost function we built to optimize the continuous particle velocity, as well as sound pressure, on the boundary of the region. In contrast to the conventional spatial sound field reproduction methods in the modal domain, the proposed method allows for non-uniform loudspeaker geometry with a limited number of loudspeakers, thus providing a flexible array arrangement. The performance of the proposed method is evaluated through numerical simulations in both a free field and a reverberant room. Finally, we prove the proposed method in an objective experiment with real-world measurements of room impulse response.

Sound intensity is an acoustic quantity closely linked with human perception of sound location, and it can be controlled to create a high level of realism to humans in soundfield reproduction systems. In this article, we present an intensity matching technique to optimally reproduce sound intensity over a continuous spatial region using an irregular loudspeaker array. This avoids several known limitations in the previous works on intensity based soundfield reproduction, such as a single sweet spot for the listener and a regular loudspeaker geometry that is difficult to implement in real-world applications. In contrast to the previous works, the new technique uses a cost function we built to optimize sound intensity over space by exploiting spatial sound intensity distributions. The spatial sound intensity distribution is represented by spherical harmonic coefficients of sound pressure, which are widely used to describe a spatial soundfield. Compared to the conventional spatial soundfield reproduction method of pressure matching in the spherical harmonic domain and the HOA max-rE decoding method optimizing sound intensity at a single position, we show that the intensity matching technique has better overall performance with two different irregular loudspeaker layouts through simulations. The impact of microphone noise on reproduction performance is also assessed. Finally, we carry out perceptual localization experiments to validate the proposed method.

The localization of acoustic sound sources is beneficial to signal processing applications of speech enhancement, dereverberation, separation and tracking. Difficulties in position estimation arise in real world environments due to coherent reflections degrading performance of subspace localization techniques. This paper proposes a method of multiple signal classification (MUSIC) subspace localization, which is suitable for reverberant rooms. The method is based on the modal decomposition of a room’s region-to-region transfer function, which is assumed to be known. We perform a numerical simulation of four sound sources in a reverberant room, and show that the localization method exhibits increased spatial resolution and distance focusing abilities when the region-to-region transfer function is incorporated. The proposed method is successful in estimating three-dimensional sound source positions without distortion due to reverberation.

The accuracy and perception of soundfields produced by loudspeaker arrays are strongly influenced by the inherent characteristics of the commercial loudspeakers. This paper analyzes such characteristics of loudspeakers by deriving equivalent theoretical models, and by studying their impact on soundfield reproduction. A number of acoustic models are investigated, including plane waves decomposition, point source decomposition and mixed source decomposition. Each proposed model employs three effective sparse decomposition algorithms for optimized solutions, including iteratively reweighted least squares (IRLS), matching pursuit (MP) and least absolute shrinkage and selection operator (LASSO). A successful model shall enable the prediction of the soundfield outside the original recording region. Therefore, we validate the effectiveness of the models by comparing the simulated soundfield with secondary measurements obtained beyond the original area. Experimental results have confirmed that both the plane wave and mixed source model achieve promising performance with respect to the proposed metrics.

Active Noise Cancellation (ANC) is a well researched topic for minimizing unwanted acoustic noise, and spatial ANC is a recently introduced concept that focuses on continuous spatial regions. Adaptive filter designing for spatial ANC is often based on frequency-domain spherical harmonic decomposition method, which has a major limitation due to the increased system latency. In this paper, we develop a time-domain spherical harmonic based signal decomposition method and use it to develop two time-space domain feed-forward adaptive filters for spatial ANC. Through simulations we show that the proposed methods can achieve higher noise reduction performance over the control region with microphones located on the surface of the region compared to the conventional time-domain adaptive filter.

Multizone reproduction has been focused on reproducing sounds in an empty listening space. However, there are always scatterers such as human heads in sound zones, generating scattered sound fields and causing degraded system performance. In this work, we develop a modal-domain method for 2.5D multizone reproduction with a solid object in the bright zone. Analytical expressions of the incident and scattered fields are developed. We then propose an active control strategy to correct the scattering effect. In the reproduction stage, we use the weighted mode matching approach to achieve the optimal control over the entire region. Simulation results show that in comparison with the conventional method which does not consider the scattering effect, the proposed method can achieve higher acoustic contrast performance over a broadband frequency range.

In this paper, we propose a planar array of dipole (or first-order) loudspeakers to reproduce a full three dimensional (3D) exterior soundfield over a desired spatial region. The proposed method is inspired by the spherical harmonics based solution for 3D soundfield reproduction. When decomposed in terms of spherical harmonics, we separate the corresponding soundfield coefficients into two sets (even and odd) and exploit the inherent properties of Legendre polynomials to control each set using a planar array of dipole (or first-order) speakers. We provide simulation examples to demonstrate the performance of the proposed method.

One major issue of implementing broadband active noise control systems in reverberant rooms is the lack of reference signals. In this work, by exploiting the spatial sound field characteristics, a time-domain sound field separation method is developed to generate the reference signal for broadband active noise control systems in reverberant rooms. The time-domain sound field separation method separates the outgoing field produced by the primary source from the secondary source feedback and room reverberation on a spherical array, based on spherical harmonic decomposition of the sound field and the radial particle velocity measured by the array. Both the effectiveness of the time-domain sound field separation method and the applicability of the separated outgoing field as the reference signal for a broadband active noise control system in a reverberant room are demonstrated by simulations.

Active noise control (ANC) over a sizeable space requires a large number of reference and error microphones to satisfy the spatial Nyquist sampling criterion, which limits the feasibility of practical realization of such systems. This paper proposes a mode-domain feedforward ANC method to attenuate the noise field over a large space while reducing the number of microphones required. We adopt a sparse reference signal representation to precisely calculate the reference mode coefficients. The proposed system consists of circular reference and error microphone arrays, which capture the reference noise signal and residual error signal, respectively, and a circular loudspeaker array to drive the anti-noise signal. Experimental results indicate that above the spatial Nyquist frequency, our proposed method can perform well compared to a conventional methods. Moreover, the proposed method can even reduce the number of reference microphones while achieving better noise attenuation.

The mode matching based multizone reproduction has mainly been focused on a purely 2D theory which is inadequate to fit the 3D reality. Its extension to the 3D theory however requires many secondary sources and a high computational complexity. In this paper, a weighted mode matching approach is developed for 2.5D multizone reproduction. The multizone soundfield is reproduced in the horizontal plane within a circular control region using the loudspeakers modelled as 3D point sources. We propose weighting the Bessel-spherical harmonic modes for 2.5D reproduction and a matching between the desired and reproduced soundfields over the entire control region. Simulation results show that in comparison with the conventional 2.5D reproduction method a more accurate reproduction is achieved using the proposed weighting approach.

We proposed a spatial coherence-based PSD estimation and source separation technique in [1] using a 32-channel spherical microphone array. While the proposed spherical microphone-based method exhibited a satisfactory performance in separating multiple sound sources in a reverberant environment, the use of a large number of microphones remains an issue for some practical considerations. In this paper, we investigate an alternative array structure to achieve spatial coherence-based source separation using a planar microphone array. This method is particularly useful in separating a limited number of sound sources in a mixed acoustic scene. The simplified array structure we used here can easily be integrated with many commercial acoustical instruments such as smart home devices to achieve better speech enhancements.

The fifth generation of mobile communications (5G) is expected to exploit the concept of location-aware communication systems. Therefore, there is a need to understand the localization limits in these networks, particularly, using millimeter-wave technology (mmWave). Contributing to this understanding, we consider single-anchor localization limits in terms of 3D position and orientation error bounds for mmWave multipath channels, for both the uplink and downlink. It is found that uplink localization is sensitive to the orientation angle of the user equipment (UE), whereas downlink is not. Moreover, in the considered outdoor scenarios, reflected and scattered paths generally improve localization. Finally, using detailed numerical simulations, we show that mmWave systems are in theory capable of localizing a UE with sub-meter position error, and sub-degree orientation error.

In this paper, we propose an efficient technique for estimating individual power spectral density (PSD) components, i.e., PSD of each desired sound source as well as of noise and reverberation, in a multisource reverberant sound scene with coherent background noise. We formulate the problem in the spherical harmonics domain to take the advantage of the inherent orthogonality of the spherical harmonics basis functions and extract the PSD components from the cross-correlation between the different sound field modes. We also investigate an implementation issue that occurs at the nulls of the Bessel functions and offer an engineering solution. The performance evaluation takes place in a practical environment with a commercial microphone array in order to measure the robustness of the proposed algorithm against all the deviations incurred in practice. We also exhibit an application of the proposed PSD estimator through a source septation algorithm and compare the performance with a contemporary method in terms of different objective measures.

Noise control and cancellation over a spatial region is a fundamental problem in acoustic signal processing. In this paper, we utilize wave-domain adaptive algorithms to iteratively calculate the secondary source driving signals and to cancel the primary noise field over the control region. We propose wave-domain active noise control algorithms based on two minimization problems: first, minimizing the wave-domain residual signal coefficients, and second, minimizing the acoustic potential energy over the region, and derive the update equations with respect to two variables, the loudspeaker weights and wave-domain secondary source coefficients. Simulation results demonstrate the effectiveness of the proposed algorithms, more specifically the convergence speed and the noise cancellation performance in terms of the noise reduction level and acoustic potential energy reduction level over the entire spatial region.

Due to strong interchannel interference in multichannel active noise control (ANC), there are fundamental problems associated with the filter adaptation and online secondary path modeling remains a major challenge. This paper proposes a wave-domain adaptation algorithm for multichannel ANC with online secondary path modelling to cancel tonal noise over an extended region of two-dimensional plane in a reverberant room. The design is based on exploiting the diagonal-dominance property of the secondary path in the wave domain. The proposed wave-domain secondary path model is applicable to both concentric and nonconcentric circular loudspeakers and microphone array placement, and is also robust against array positioning errors. Normalized least mean squares-type algorithms are adopted for adaptive feedback control. Computational complexity is analyzed and compared with the conventional time-domain and frequency-domain multichannel ANCs. Through simulation-based verification in comparison with existing methods, the proposed algorithm demonstrates more efficient adaptation with low-level auxiliary noise.

This paper proposes a new channel training (CT) scheme for a full-duplex receiver to enhance physical layer security. Equipped with NBfull-duplex antennas, the receiver simultaneously receives the information signal and transmits artificial noise (AN). In order to reduce the non-cancellable self-interference due to the transmitted AN, the receiver has to estimate the self-interference channel prior to the data communication phase. In the proposed CT scheme, the receiver transmits a limited number of pilot symbols that are known only to itself. Such a secret CT scheme prevents an eavesdropper from estimating the jamming channel from the receiver to the eavesdropper, hence effectively degrading the eavesdropping capability. We analytically examine the connection probability (i.e., the probability of the data being successfully decoded by the receiver) of the legitimate channel and the secrecy outage probability due to eavesdropping for the proposed secret CT scheme. Based on our analysis, the optimal power allocation between CT and data/AN transmission at the legitimate transmitter/receiver is determined. Our examination shows that the newly proposed secret CT scheme significantly outperforms the non-secret CT scheme that uses publicly known pilots when the number of antennas at the eavesdropper is larger than one.

Location-aware communication systems are expected to play a pivotal part in the next generation of mobile communication networks. Therefore, there is a need to understand the localization limits in these networks, particularly, using millimeter-wave technology (mm-wave). Towards that, we address the uplink and downlink localization limits in terms of 3D position and orientation error bounds for mm-wave multipath channels. We also carry out a detailed analysis of the dependence of the bounds on different system parameters. Our key findings indicate that the uplink and downlink behave differently in two distinct ways. First of all, the error bounds have different scaling factors with respect to the number of antennas in the uplink and downlink. Secondly, uplink localization is sensitive to the orientation angle of the user equipment (UE), whereas downlink is not. Moreover, in the considered outdoor scenarios, the non-line-of-sight paths generally improve localization when a line-of-sight path exists. Finally, our numerical results show that mm-wave systems are capable of localizing a UE with sub-meter position error, and sub-degree orientation error.

Distributed image fusion over networks has had little coverage in the literature, particularly considering the recent emergence of large networks of imaging sensors such as radio telescope arrays and wireless self-contained node networks. We present a fully asynchronous and distributed approach for image fusion in a general network with partially overlapping node fields of view. We use the example of an aerial surveillance drone network to present the advantages of our system and show that the communication power required for performing image fusion in-network is orders of magnitude lower than transmitting all raw images back to a distant central processor, while still achieving the same fusion performance.

This paper presents an experimental study on a novel technique to blindly estimate the directional properties of room reflections using a spherical microphone array. The algorithm is developed based on a spatial correlation model formulated in the spherical harmonics domain. This model expresses the cross correlation matrix of the recorded soundfield coefficients in terms of direct sound and reflections. The directional gain of the reflected path is estimated from the above model, which provides information on the DOAs of dominant wall reflections. The practical feasibility of the proposed algorithm is evaluated using a subset of the speech corpus from the ACE (Acoustic Characterization of Environments) Challenge.

The performance of an ANC system largely depends on the availability of an accurate secondary path model. This is however a major challenge in multichannel ANC where the computational complexity increases significantly with the number of secondary sources and error sensors. This paper proposes wave-domain adaptive processing algorithm for multichannel ANC with online secondary modelling to cancel a tonal noise over a region of space within a reverberant room. The design is based on exploiting a special property of the secondary path model in the wave domain. A feedback control system is implemented, where a single microphone array is placed at the boundary of the control region to measure the residual signals, and a loudspeaker array reproduces secondary sources to generate the anti-noise signals and auxiliary noise for secondary path modelling. Through experimental verification in comparison with existing methods the proposed algorithm demonstrates more efficient adaptation with low-level auxiliary noise.

In this work, we propose a new channel training (CT) scheme to enhance physical layer security in a full-duplex wiretap channel, where the multi-antenna and full-duplex receiver simultaneously receives the information signal and transmits artificial noise (AN). In order to suppress the self-interference caused by AN, the receiver has to estimate the self-interference channel prior to the data communication phase. In the proposed CT scheme, the receiver transmits limited pilot symbols which are known only to itself, which prevents the eavesdropper from estimating the jamming channel from the receiver to the eavesdropper, hence effectively degrades the eavesdropping capability. Compared with the traditional CT scheme that uses publicly known pilots, the newly proposed secret CT scheme offers significantly better performance when the number of antennas at the eavesdropper is larger than one, e.g., Ne> 1. The optimal power allocation between CT and data/AN transmission at the legitimate transmitter/receiver is determined for the proposed secret CT scheme.

With the growing popularity of ad‐hoc sensor networks, spatially distributed multiple‐input multiple‐output (MIMO) systems have drawn a lot of attention. This work considers a spatially distributed MIMO system with randomly distributed transmit and receive antennas over spatial regions. The authors use the modal decomposition of wave propagation to analyse the performance limits of such system, since the sampling of the spatial regions populated with antennas is a form of mode excitation. Specifically, they decompose signals into orthogonal spatial modes and apply concepts of MIMO communications to quantify the instantaneous capacity and the outage probability. The authors’ analysis shows that analogous to conventional point‐to‐point MIMO system, the instantaneous capacity of spatially distributed MIMO system over Rayleigh fading channel is equivalent to a Gaussian random variable. Afterwards, they derive an accurate closed‐form expression for the outage probability of proposed system utilising the definition of instantaneous capacity. Besides, in rich scattering environment, the spatially distributed MIMO system provides best performance when the spatial regions are of same size, and each region is equipped with equal number of antennas. Furthermore, to facilitate the total transmit power allocation among the channels, they propose an algorithm which indicates a significant performance improvement over conventional equal transmit power allocation scheme, even at low signal‐to‐noise ratio.

The direct-to-reverberant ratio (DRR) is an important characterization of a reverberant environment. This paper presents a novel blind DRR estimation method based on the coherence function between the sound pressure and particle velocity at a point. First, a general expression of coherence function and DRR is derived in the spherical harmonic domain, without imposing assumptions on the reverberation. In this paper, DRR is expressed in terms of the coherence function as well as two parameters that are related to statistical characteristics of the reverberant environment. Then, a method to estimate the values of these two parameters using a microphone system capable of capturing first-order spherical harmonics is proposed, under three assumptions which are more realistic than the diffuse field model. Furthermore, a theoretical analysis on the use of plane wave model for direct path signal and its effect on DRR estimation is presented, and a rule of thumb is provided for determining whether the point source model should be used for the direct path signal. Finally, the ACE challenge dataset is used to validate the proposed DRR estimation method. The results show that the average full band estimation error is within 2 dB, with no clear trend of bias.

The direct-to-reverberant ratio (DRR), which describes the energy ratio between the direct and reverberant component of a soundfield, is an important parameter in many audio applications. In this paper, we present a multichannel algorithm, which utilizes the blind recordings of a spherical microphone array to estimate the DRR of interest. The algorithm is developed based on a spatial correlation model formulated in the spherical harmonics domain. This model expresses the cross correlation matrix of the recorded soundfield coefficients in terms of two spatial correlation matrices, one for direct sound and the other for reverberation. While the direct path arrives from the source, the reverberant path is considered to be a nondiffuse soundfield with varying directional gains. The direct and reverberant sound energies are estimated from the aforementioned spatial correlation model, which then leads to the DRR estimation. The practical feasibility of the proposed algorithm was evaluated using the speech corpus of the acoustic characterization of environments challenge. The experimental results revealed that the proposed method was able to effectively estimate the DRR of a large collection of reverberant speech recordings including various environmental noise types, room types and speakers.

The utilization of the millimeter-wave frequency band (mm-wave) in the fifth generation ({5G}) of mobile communication is a highly-debated current topic. Mm-wave MIMO systems will use arrays with large number of antennas at the transmitter and the receiver, implemented on a relatively small area. With the inherent high directivity of these arrays, algorithms to help the user equipment find the base station and establish a communication link should be carefully designed. Towards that, we examine two beamforming schemes, namely, random-phase beamforming (RPBF) and directional beamforming (DBF), and test their impact on the Cram\'er-Rao lower bounds (CRB) of jointly estimating the direction-of-arrival, direction-of-departure, time-of-arrival, and the complex channel gain, under the line-of-sight channel model. The results show that the application of RPBF is more appropriate in the considered scenario as it attains a lower CRB with fewer beams compared to DBF.

We examine for the first time the impact of transmitter-side correlation on the secure transmission with artificial noise (AN), based on which a new power allocation strategy for AN is devised for physical layer security enhancement. Specifically, we design a correlation-based power allocation (CPA) for AN, of which the optimality in terms of achieving the minimum secrecy outage probability is analytically proved in the large system regime with the number of transmit antennas approaching infinity. Our numerical results demonstrate that CPA is nearly optimal and can significantly outperform the widely-used uniform power allocation (UPA) even for a moderate (finite) number of correlated transmit antennas. Our numerical results also reveal a fundamental difference between the secrecy performance of CPA and that of UPA. When the number of correlated transmit antennas increases, we find that the secrecy outage probability of CPA always reduces while the secrecy outage probability of UPA suffers from a saturation point.

Until now, distributed acoustic beamforming has focused on optimizing for a beamformer over an entire network, with each node contributing to the beamformer output. We present a novel approach that introduces sparsity to this beamformer computation, where we attempt to optimize for a subset of nodes within the network that produce SNR gains roughly equivalent to that of the optimal MVDR case. Due to the physical nature of sound, this approach trades a small loss in SNR for a large reduction in communication power and iterations required to produce a beamformer output by reducing the active node set of our network. Our approach operates in a fully distributed and asynchronous manner and does not require a high update iteration rate to produce an output at each sample.

The performance of binaural speech source localization systems can be significantly impacted by an imperfect selection of spatial localization cues, due to the limited bandwidth of speech, and the effects of noise. In order to mitigate these impacts, this paper presents a novel method that combines a deterministic localization approach with a spatial feature learning process. Here, we (i) obtain a composite feature vector derived from analysing the mutual information between different spatial cues and (ii) estimate the optimum feature combination that minimizes the angular localization error in three dimensional space. The performance of the proposed mutual information based feature learning approach is evaluated for a range of speech inputs and noise conditions. We also demonstrate that the proposed approach improves the localization accuracy and its robustness, with respect to traditional localization algorithms, especially in the relatively low signal-to-noise ratio localization scenarios.

In this paper, we investigate active noise control over large 2D spatial regions when the noise source is sparsely distributed. The l1relaxation technique originated from compressive sensing is adopted and based on that we develop the algorithm for two cases: multipoint noise cancellation and wave domain noise cancellation. This results in two new variants (i) zero-attracting multi-point complex FxLMS and (ii) zero-attracting wave domain complex FxLMS. Both approaches use a feedback control system, where a microphone array is distributed over the boundary of the control region to measure the residual noise signals and a loudspeaker array is placed outside the microphone array to generate the anti-noise signals. Simulation results demonstrate the performance and advantages of the proposed methods in terms of convergence rate and spatial noise reduction levels.

Dimensionality of spatio-temporal signals is fundamental to the performance limits of any wireless communications system with multiple antennas. In this paper, we use the spherical harmonic analysis of wave propagation to derive a closed-form expression for the dimensionality of band-limited wireless signals observed over a finite region of space within a finite time window. Specifically, we decompose spatio-temporal signals into independent and orthogonal spatial modes (waveforms) and apply Shannon's classical signal dimensionality result, namely, time-frequency bandwidth product to each spatial mode. Our analysis shows that: 1) the effective observation time of all the spatial modes is the sum of the finite time window and the time it takes a wave front to propagate across the entire spatial region and 2) though the effective bandwidth at lower spatial modes is equal to the bandwidth of the underlying transmitted signal, the effective bandwidth decreases as the mode index increases depending on the acceptable signal-to-noise ratio at each spatial mode. Thus, only a finite number of spatial modes (independent channels) carry information with its own time-frequency bandwidth product. These findings show that the classical signal dimensionality result of time-bandwidth product does not extend directly to the product of spatial degrees of freedom and time-bandwidth product. Simulation result indicates that increasing the radius of the region leads to a sub-quadratic growth in the derived signal dimensionality irrespective of the bandwidth or the observation time.

This paper, for the first time, examines the impact of transmitter-side correlation on the artificial-noise (AN)-aided secure transmission, based on which a new power allocation strategy for AN is devised for physical layer security enhancement. Specifically, we design a correlation-based power allocation (CPA) for AN, of which the optimality in terms of achieving the minimum secrecy outage probability is analytically proved in the large system regime with the number of transmit antennas approaching infinity. In order to fully reveal the benefits of the CPA, we derive easy-to-evaluate expressions for the secrecy outage probability achieved by the CPA. Our study demonstrates that the CPA is nearly optimal and significantly outperforms the widely used uniform power allocation (UPA) even for a moderately small number of correlated transmit antennas. Furthermore, our numerical results reveal a fundamental difference between the CPA and UPA. That is when the number of correlated transmit antennas increases, the secrecy outage probability of the CPA always reduces while the secrecy outage probability of the UPA suffers from a saturation point.

Binaural localization of speech sources in 3-D, using head-related transfer functions (HRTFs), always suffers elevation ambiguity due to the limited high frequency spectral information available at the receivers. This paper presents a method that overcomes this limitation by exploiting the interaural phase and magnitude features present in the HRTF. We (i) introduce a new feature vector that combines these two sets of features in a non-linear fashion, and (ii) propose a mechanism to extract this feature vector free from distortion by the speech spectra. The performance of the proposed method is evaluated and compared with a correlation-based HRTF database matching approach and a two-step localization technique for multiple source positions, HRTFs (individuals) and speech inputs. The results suggest that up to 20% improvement in localization performance can be achieved for moderate signal-to-noise ratios.

This paper proposes an efficient parameterization of the room transfer function (RTF). Typically, the RTF rapidly varies with varying source and receiver positions, hence requires an impractical number of point to point measurements to characterize a given room. Therefore, we derive a novel RTF parameterization that is robust to both receiver and source variations with the following salient features: 1) The parameterization is given in terms of a modal expansion of 3D basis functions. 2) The aforementioned modal expansion can be truncated at a finite number of modes given that the source and receiver locations are from two sizeable spatial regions, which are arbitrarily distributed. 3) The parameter weights/coefficients are independent of the source/receiver positions. Therefore, a finite set of coefficients is shown to be capable of accurately calculating the RTF between any two arbitrary points from a pre-defined spatial region where the source(s) lie and a pre-defined spatial region where the receiver(s) lie. A practical method to measure the RTF coefficients is also provided, which only requires a single microphone unit and a single loudspeaker unit, given that the room characteristics remain stationary over time. The accuracy of the above parameterization is verified using appropriate simulation examples.

This paper considers a functional modeling of a head-related transfer function (HRTF) where the spatial-portion is constrained to be expanded using spherical harmonics and the frequency-portion is expanded in terms of standard closed-form orthonormal functions, which may be user selected. We derive an objective evaluation metric to compare the relative energy efficiencies of candidate functional models using empirical HRTF database measurements and robust estimation techniques. Among four sets of closed-form orthonormal functions the complex exponentials are identified as the most efficient to represent the frequency-portion in the spherical harmonics-based HRTF functional model. The proposed model is evaluated across three HRTF data sets: 1) CIPIC database, 2) the MIT KEMAR (Knowles Electronics Mannequin for Acoustics Research) database, and 3) the ANU KEMAR database.

The existing research on physical layer security commonly assumes the number of eavesdropper antennas to be known. Although this assumption allows one to easily compute the achievable secrecy rate, it can hardly be realized in practice. In this paper, we provide an innovative approach to studying secure communication systems without knowing the number of eavesdropper antennas by introducing the concept of spatial constraint into physical layer security. Specifically, the eavesdropper is assumed to have a limited spatial region to place (possibly an infinite number of) antennas. From a practical point of view, knowing the spatial constraint of the eavesdropper is much easier than knowing the number of eavesdropper antennas. We derive the achievable secrecy rates of the spatially-constrained system with and without friendly jamming. We show that a non-zero secrecy rate is achievable with the help of a friendly jammer, even if the eavesdropper places an infinite number of antennas in its spatial region. Furthermore, we find that the achievable secrecy rate does not monotonically increase with the jamming power, and hence, we obtain the closed-form solution of the optimal jamming power that maximizes the secrecy rate.

This letter presents a Gaussian-sum cubature Kalman filter with improved robustness compared to the original algorithm proposed by the authors in, which demonstrated excellent accuracy and efficiency for the bearings-only tracking problem. Modifications are made in the splitting and merging procedure of the Gaussian components in the algorithm. Simulation results confirm the improved robustness of the modified algorithm against the choice of threshold level for the splitting procedure.

Three dimensional sound field reproduction based on higher order Ambisonics requires the placement of loudspeakers on a sphere that surrounds the target reproduction region. The deployment of a spherical array is not trivial especially for implementation in real rooms where the placement flexibility is highly desirable. This paper proposes a design of multiple circular loudspeaker arrays for reproducing three dimensional sound fields originating from a limited region of interest. We apply a functional analysis framework to formulate the sound field reproduction problem in a closed form. Secondary source distributions and target sound fields are modeled as two Hilbert spaces and mapped by an integral operator and its adjoint operator, from which a self-adjoint operator is constructed and the singular value decomposition is applied to represent source distributions and sound fields with two sets of interrelated singular functions. We derive the solutions for a circular secondary source arrangement and propose the design of placing multiple circular loudspeaker arrays only over the limited region of interest. Such a design allows for non-spherical and non-uniform loudspeaker placement and thus provides a flexible array arrangement. The reproduction accuracy of the proposed method is verified through numerical simulations.

Successful recording of large spatial soundfields is a prevailing challenge in acoustic signal processing due to the enormous numbers of microphones required. This paper presents the design and analysis of an array of higher order microphones that uses 2D wavefield translation to provide a mode matching solution to the height invariant recording problem. It is shown that the use of Mth order microphones significantly reduces the number of microphone units by a factor of 1/(2M + 1) at the expense of increased complexity at each microphone unit. Robustness of the proposed array is also analyzed based on the condition number of the translation matrix while discussing array configurations that result in low condition numbers. The white-noise gain (WNG) of the array is then derived to verify that improved WNG can be achieved when the translation matrix is well conditioned. Furthermore, the array's performance is studied for interior soundfield recording as well as exterior soundfield recording using appropriate simulation examples.

Mitigating the effects of reverberation is a significant challenge for real-world spatial soundfield reproduction, but the necessity of a large number of reproduction channels increases the complexity and presents several challenges to existing listening room compensation techniques. In this paper, we present an adaptive room compensation method to overcome the effects of reverberation within a region, using a model description of the reverberant soundfield. We propose the reverberant channel estimation and compensation be carried out in a single step using completely decoupled adaptive filters; thus, reducing the complexity of the overall process. We compare the soundfield reproduction performance with existing adaptive and nonadaptive room compensation methods through several simulation examples. The performance of the proposed method is comparable to existing techniques, and achieves a normalized wideband region reproduction error of 1% at a signal-to-noise ratio of 50 dB, within a 1 m radius region of interest using 60 loudspeakers and 55 microphones at frequencies below 1 kHz. Robust behavior of the room compensator is demonstrated down to direct-to-reverberant-path power ratios of -5 dB. Overall, the results suggest that the proposed method can diagonalize the room compensation system, leading to a more robust and parallel implementation for spatial soundfield reproduction.

This paper presents a track-before-detect algorithm based on the particle filter for simultaneously detecting and tracking a single target with nearly constant velocity. The measurement data available to the algorithm is the raw direction-of-arrival spectrum of the received wideband signals at a single observation platform. The proposed method does not require prior information on the target existence probability and it is capable of detecting a target of very low signal-to-noise ratio (-30dB). Simulation results show that the proposed algorithm demonstrates good performance in detecting and tracking a target with reasonably low errors in the estimated target states over the observation period.

Three dimensional surround sound reproduction over large areas is a prevailing challenge due to the enormous numbers of loudspeakers required. In this paper, we propose an array of higher order loudspeakers which provide a mode matching solution to the problem based on 3D wavefield translation. It is shown that for a given bandwidth, the use of Lthorder sources significantly brings down the minimum loudspeaker requirement by a factor of 1=(L + 1)2. Furthermore, the array is shown to be capable of exterior field cancellation, increasing its performance in echoing environments. Design examples are given for interior field, exterior field and interior and exterior combined field reproduction.

The equalization of reverberation effects is essential for spatial soundfield reproduction, but estimation of the reverberant channel presents several challenges to existing equalization techniques. This paper presents a method of active acoustic echo cancellation (AEC) for soundfield reproduction applications, using a modal description of the reverberant soundfield. We describe how individual modes of the measured soundfield can be equalized adaptively, thus reducing the complexity of the channel estimation process. AEC and reproduction performance is compared with existing adaptive and non-adaptive equalization techniques through simulation examples. Equalization performance is comparable to existing methods, achieving a normalized region reproduction error of 1% and echo return loss enhancement of 15 - 30 dB at 50 dB SNR. The results suggest that the proposed model can be used to obtain a parallel implementation of a room equalizer for active AEC.

Sensor arrays mounted on complex-shaped rigid bodies are a common feature in many practical broadband direction of arrival (DOA) estimation applications. The scattering and reflections caused by these rigid bodies introduce complexity and diversity in the frequency domain of the channel transfer function, which presents several challenges to existing broadband DOA estimators. This paper presents a novel high resolution broadband DOA estimation technique based on signal subspace decomposition. We describe how broadband signals can be decomposed into narrow subband components, and combined such that the frequency domain diversity is retained. The DOA estimation performance is compared with existing techniques using a uniform circular array and a sensor array on a hypothetical rigid body. An improvement in closely spaced source resolution of up to 6 dB is observed for the sensor array on the hypothetical rigid body, in comparison to the uniform circular array. The results suggest that frequency domain diversity, introduced by complex-shaped rigid bodies, can provide higher resolution and clearer separation of closely spaced broadband sound sources.

The implementation of 3D sound reproduction is well founded theoretically, but the requirements for the number of loudspeakers and array geometry make it impractical using conventional technology. In practice, a 2D array of loudspeakers is commonly used which restricts the reproduction of sources to those in the horizontal plane and requires a restricted form for the free-field loudspeaker driving signals based on the sectorial spherical harmonics. However, reflections can impair reproduction quality when using the free-field solution. Since first-order loudspeakers are commercially available which increase the direct to reflected sound ratio, we extend the sectorial solution for the loudspeaker excitation signals to the first-order case. We also investigate the reproduction accuracy for both zeroth and first-order loudspeakers using numerical simulations.

Similar to computed tomography, the images from photoacoustic tomography are piecewise smooth and reconstructions can be enhanced by applying Total Variation (TV) minimization. Also, the photoacoustic image must satisfy both the data fidelity and a non negativity constraint. This paper proposes a constrained TV minimization formulation in the Fourier Bessel domain which has the advantage of reducing the dimension of the fidelity constraint and the projection onto this constraint can be performed by a simple projection onto the l2ball. The TV minimization post processing improves the quality of images reconstructed and recovers the lost, low frequency information when sensor lower frequency limit is higher than zero.

Transmission of signals using multiple antennas can significantly improve the energy efficiency of a wireless network, and the proper alignment of the transmitted signals' phases at the receiver is one of the key factors so this efficiency improvement can be realized. In a time-varying channel, due to the relative motion between the transmitters and the receiver, the development of a scheme that guarantees such alignment is very challenging. In this paper, considering a distributed transmit beamforming scenario, an algorithm to achieve such phase alignment of signals in a time-varying channel is proposed. A simple formula is derived, which can be adopted by each transmitter to compute its beamforming weight's phase, and it is shown that the use of this perturbation results in a significant improvement in terms of speed of convergence at the receiver. With simulation it is shown that, using the proposed scheme, the transmitted signals' phase aligned at the receiver 33% faster than the one-bit feedback scheme. The average theoretical bit error rate is achieved 50% faster relative to the one-bit feedback scheme with on average 18% less number of antennas at the transmitter.

This study proposes a framework for calculating level-crossing rate and average fade duration in wireless communication systems operating in Rayleigh fading channel environments. The complexity of the calculations involved is greatly reduced compared to the conventional way of calculating these two measures. To demonstrate the strength of the proposed framework, the authors applied it to two-dimensional and three-dimensional isotropic and non-isotropic scattering environments and the results are compared to available results in the literature.

This paper deals with two important questions associated with HRTF measurement: 1) “what is the required angular resolution?,” and 2) “what is the most suitable sampling scheme?.” The paper shows that a well-defined finite number of spherical harmonics can capture the head-related transfer function (HRTF) spatial variations in sufficient detail, which is defined as the HRTF spatial dimensionality. For the 20-kHz audible frequency range, the value of the dimensionality means a high-directional resolution HRTF measurement is required. Considering such a high-resolution measurement, a number of sampling criteria have been identified from both mechanical setup and data processing aspects. Different sampling candidates are then compared to demonstrate that the best method which satisfies all requirements is the class termed as IGLOO. A fast spherical harmonic transform algorithm based on the IGLOO scheme is developed to accelerate the high-resolution data analysis. The proposed method is validated through simulation and experimental data acquired from a KEMAR mannequin.

We propose a novel algorithm for temporal tracking of the fibrillatory frequency for Atrial Fibrillation (AF) ECG. Both Atrial activity extraction and fibrillatory frequency tracking are combined into a single algorithm and analytical expressions are derived for final time-frequency distributions, thus the proposed method is computationally efficient. It can also serve the general purpose of time-frequency analysis of a non stationary signal with missing or corrupted data segments. The method uses short-time expansion of an orthogonal basis set, and regularized least squares solution used to compute a stable coefficients for the basis. A simplified and computationally efficient time-frequency distribution is derived based only on the coefficient vector. The algorithm is successfully applied to simulated as well as real ECG with AF, and its ad vantages over conventional average beat subtraction method are discussed.

Sound reproduction systems aim to produce a desired sound field over a region of space. At high frequencies, the number of loudspeakers required is prohibitive. This paper shows that the use of Nth order loudspeakers, in which each loudspeaker produces polar responses up to cos(Nφ) and sin (Nφ) , produces accurate reproduction over N times the area of a first order array and can largely eliminate any exterior field. This allows a significant reduction in the number of loudspeaker units, at the expense of increased complexity in each loudspeaker unit.

This paper consider the problem of how to evaluate the efficiency of a 3D continuous functional HRTF model in rep resenting measured data. The proposed method is based on Karhunen-Loeve theorem. After investigating the optimization problem of representing the process with an finite linear combination of orthogonal functions in L space by means of the least squared method, the variance of random variables involved in the model is found the key metric in efficiency evaluation of a given functional model. Then, the coefficients of the 3D continuous HRTF model are analyzed. The results show that the efficiency of the model in spatial component expansion is around 70% and the best choice in frequency component expansion is the spherical Bessel function.

This paper derives closed form expressions for second order statistics of mobile-to-mobile wireless channels in non isotropic environments. Specifically, we consider Rice, Rayleigh and Nakagami fading channels in four different non isotropic scattering environments with Angle of Departure (AoD) and Angle of Arrival (AoA) distributions given by (i) separable Truncated Gaussian, (ii) separable von-Mises, (iii) Truncated Gaussian bivariate and (iv) Truncated Laplacian bivariate distributions. We have shown that the second order statistics, namely, the Level Crossing Rate (LCR) and the Average Fade Duration (AFD) in different fading channels can be expressed in terms of known scattering coefficients of the above angle of departure and arrival distributions. We have illustrated the usefulness of the results by plotting LCR and AFD against the channel correlation for various scattering scenarios.

Three-dimensional spatial sound field reproduction enables enhanced immersive acoustic experience for a listener. Recreating an arbitrary 3-D spatial sound field using a practically realizable array of loudspeakers is a challenging problem in acoustic signal processing. This paper exploits the underlying characteristics of wavefield propagation to devise a strategy for accurate 3-D sound field reproduction inside a 3-D region of interest with practical array geometries. Specifically, we use the properties of the associated Legendre functions and the spherical Hankel functions, which are part of the solution to the wave equation in spherical coordinates, for loudspeaker placement on a set of multiple circular arrays and provide a technique for spherical harmonic mode-selection to control the reproduced sound field. We also analyze the artifacts of spatial aliasing due to the use of discrete loudspeaker arrays in the region of interest. As an illustration, we design a third-order reproduction system to operate at a frequency of 500 Hz with 18 loudspeakers arranged in a practically realizable configuration.

Spatial multizone soundfield reproduction over an extended region of open space is a complex and challenging problem in acoustic signal processing. In this paper, we provide a framework to recreate 2-D spatial multizone soundfields using a single array of loudspeakers which encompasses all spatial regions of interest. The reproduction is based on the derivation of an equivalent global soundfield consisting of a number of individual multizone soundfields. This is achieved by using spatial harmonic coefficients translation between coordinate systems. A multizone soundfield reproduction problem is then reduced to the reproduction over the entire region. An important advantage of this approach is the full use of the available dimensionality of the soundfield. This paper provides quantitative performances of a 2-D multizone system and reveals some fundamental limits on 2-D multizone soundfield reproduction. The extensions of the multizone soundfield reproduction design in reverberant rooms are also included.

This paper presents a double sided conical sensor array for common array processing applications. We show that double sided cone arrays are specially suited for extraction of spherical harmonic components of three dimensional spatial wavefields. We use a less known orthogonal relationship of spherical Bessel functions to develop a novel analysis equation for spherical harmonic extraction. As an illustration, we design and simulate a 5th order spherical harmonic decomposition array using 89 microphones to operate over a frequency band of an octave.

Spherical harmonic decomposition of wavefields is not only an active problem in acoustic signal processing but also a useful tool in a plethora of applications such as 3-D beamforming, direction of arrival estimation, and spatial sound recording. This paper presents a novel array structure consisting of a set of parallel circular arrays of sensors to decompose a wavefield into spherical harmonic components. The new structure presented here provides an alternative design to the traditional spherical microphone arrays with increased flexibility on sensor locations. We use the underlying structure of the wave propagation together with the properties of the associated Legendre functions and the spherical Bessel functions to develop a systematic approach to place circular arrays and construct a hybrid array. As an illustration, we design a fifth-order spherical harmonic decomposition array using 57 microphones to operate over a frequency band of an octave and compare it with a spherical array. We use computer simulations to show the performance of the array in a beamforming example.

This paper proposes a general framework for the calculation of space-frequency correlation at the receiver in time-invariant frequency selective channel environments. The proposed framework incorporates antenna locations to capture the spatial information of the channel and a general joint angle-delay power distribution to capture the frequency selectivity of the multipath channel. To demonstrate the strength of the proposed framework, we apply it on an orthogonal frequency division multiplexing (OFDM) system to obtain the space-frequency correlation between two antenna elements for two analytically derived joint angle-delay power distributions.

We present a novel theory and design for constructing microphone arrays to extract spherical harmonic components from sound fields. The proposed non-spherical array structure provides a flexible and alternative design to the traditional spherical microphone arrays with lesser restriction on sensor locations. We use the properties of the associated Legendre functions and the spherical Bessel functions to develop a systematic approach to place circular microphone arrays in three dimensions for hybrid array geometries. As an illustration, we design and simulate a fifth order spherical harmonic decomposition array using 70 microphones to operate over a frequency band of an octave.

Spatial multizone soundfield reproduction is a difficult problem, which has many potential applications. This paper provides a framework to recreate 2D spatial multizone soundfields using an array of loudspeakers. We derive the desired global soundfield by translating individual desired soundfields to a single global co-ordinate system and applying appropriate angular window functions. We reveal some of the fundamental limits of 2D multizone soundfield reproduction. We show that the ability of multizone reproduction is dependent on (i) maximum radius of multizones, (ii) window length (size, and nature), and (iii) radial distance to the furthermost zone. We illustrate the framework by designing and simulating a two dimensional two zone soundfield.

This paper proposes a continuous HRTF representation in both 3D spatial and frequency domains. The method is based on the acoustic reciprocity principle and a modal expansion of the wave equation solution to represent the HRTF variations with different variables in separate basis functions. The derived spatial basis modes can achieve HRTF near-field and far-field representation in one formulation. The HRTF frequency components are expanded using Fourier Spherical Bessel series for compact representation. The proposed model can be used to reconstruct HRTFs at any arbitrary position in space and at any frequency point from a finite number of measurements. Analytical simulated and measured HRTFs from a KEMAR are used to validate the model.

This paper proposes an efficient rate one space-frequency block code (SFBC) for multiple-input multiple-output orthogonal frequency division multiplexing (MIMO-OFDM) systems. The proposed SFBC incorporates concept of matched rotation precoding (MRP) to achieve full transmit diversity and optimal system performance for arbitrary number of transmit antennas, subcarrier interval and subcarrier grouping. The MRP exploits the inherent rotation property of SFBC and has relaxed restrictions on subcarrier interval and subcarrier grouping, making it ideal for adaptive/time varying systems. The lowerbound of the coding gain for MRP is derived and shown that it is useful when designing a SFBC for practical scenarios, e.g. when transmitters have only partial knowledge of power delay profile or when the power delay profile has only a few dominant delayed paths. Simulation results show that the MRP can achieve a similar or better performance than existing SFBCs.

Clarke's classical model of mobile-radio reception assumes isotropic-rich scattering around the mobile receiver antenna. The assumption of isotropic scattering is valid only in limited circumstances. Here, a generalised Clarke model is developed, which is applicable to mobile-radio reception in general scattering environments. The authors give expressions for the autocorrelation and power spectral density of the channel fading process and demonstrate the generality of the model by applying it to different non-isotropic scattering scenarios. Using the generalised model, the effect of mobile direction of travel and the non-isotropicity on the statistics of the channel fading process is analysed. It is also shown that if the mobile direction of travel is equiprobable in all directions, a non-isotropic scattering environment on average is as good as an isotropic scattering environment.

Reproduction of a soundfield is a fundamental problem in acoustic signal processing. A common approach is to use an array of loudspeakers to reproduce the desired field where the least-squares method is used to calculate the loudspeaker weights. However, the least-squares method involves matrix inversion which may lead to errors if the matrix is poorly conditioned. In this paper, we use the concept of theoretical continuous loudspeaker on a circle to derive the discrete loudspeaker aperture functions by avoiding matrix inversion. In addition, the aperture function obtained through continuous loudspeaker method reveals the underlying structure of the solution as a function of the desired soundfield, the loudspeaker positions, and the frequency. This concept can also be applied for the 3-D soundfield reproduction using spherical harmonics analysis with a spherical array. Results are verified through computer simulations.

This paper introduces a continuous functional model for head-related transfer functions (HRTFs) in the horizontal auditory scene. The approach uses a separable representation consisting of a Fourier-Bessel series expansion for the spectral components and a conventional Fourier series expansion for the spatial components. Being independent of the data, these two sets of basis functions remain unchanged for all subjects and measurement setups. Hence, the model can transform an individualized HRTF to a subject specific set of coefficients. A continuous functional model is also developed in the time domain. We show the efficient model performance in approximating experimental measurements by using the HRTF measurements from a KEMAR manikin and the synthetic data from the spherical head model. The statistical results are determined from a 50-subject HRTF data set. We also corroborate the predictive capability of the proposed model. The model has near optimal performance, which can be ascertained by comparison with the standard principle component analysis (PCA) and discrete Karhunen-Loeve expansion (KLE) methods at the measurement points and for a given number of parameters.

This paper provides a generalized framework to decompose a sourcefield into its spatial and frequency components using a spherical microphone array. Spherical arrays are used to decompose a soundfield into spherical harmonics which are the natural basis functions for space over directions. This paper extends this theory by including basis functions in frequency to decompose a soundfield into spatial-frequency components. Specifically, we use spherical Bessel functions as a set of basis in frequency. The paper also shows how to avoid numerical ill-conditioning inherited in open spherical arrays at frequencies corresponding to Bessel zeros.

Reproduction of a soundfield is a fundamental problem in acoustic signal processing. A common approach is to use an array of loudspeakers to reproduce the desired field where the least square method is used to calculate the loudspeaker weights. However, the least square method involves matrix inversion which may lead to errors if the matrix is poorly conditioned. In this paper, we derive a new theoretical continuous loudspeaker method to obtain the loudspeaker aperture function in order to avoid matrix inversion. In addition, the aperture function obtained through continuous loudspeaker method reveals the underlying structure of the solution as a function of the desired soundfield, the loudspeaker positions and the frequency.

Given limited or incomplete measurement data on a sphere, a new iterative algorithm is proposed on how to extrapolate signal over the whole sphere. The algorithm is based on a priori assumption that the Fourier decomposition of the signal on the sphere has finite degree of spherical harmonic coefficients, that is, the signal is mode limited. The algorithm is a simple iteration involving only the spherical harmonic decomposition. It is proven that the algorithm converges to the original signal over observation region and the convergence rate is lower bounded by the largest eigenvalue of an associated Fredholm integral equation.

In this paper, we first propose an alternative decision feedback equalizer (DFE) scheme to the DFE scheme of Labat, Macchi and Laot. Both schemes are broken into two distinct operation modes which feature a linear equalizer during acquisition and a DFE after the initial convergence has been achieved. While avoiding some potential problems with the latter DFE, our scheme achieves similar overall performance in terms of convergence speed and steady state error. A second contribution is the development of a strategy that simultaneously adapts the linear equalizer and the DFE in a parallel fashion such that the switch-over between modes is no longer necessary. Consequently, smoother and usually faster convergence is achieved as switching disruptions are reduced. This scheme is especially advantageous under noisy and severe conditions as supported by our simulations.

This paper presents a multirate space-time- frequency linear block coding scheme (STFBC) with full transmit diversity for a variety of transmission rates. The proposed multirate STFBC can achieve relatively smooth balance between the performance and the transmission rate for a given constellation size. Design of a space-time linear block coding (STBC) scheme is presented as a special case of the proposed multirate STFBC, which perform better than some of the existing STBCs. Moreover, optimized multirate STFBCs have also been compared with some of the existing STFBCs. Simulations results show that the design parameter has sufficient flexibility to achieve improved performance with reduced computational complexity.

We bound the number of electromagnetic signals which may be observed over a frequency range [F-W, F+W] a time interval [0, T] within a sphere of radius R. We show that the such constrained signals may be represented by a series expansion whose terms are bounded exponentially to zero beyond a threshold. Our result implies there is a finite amount of information which may be extracted from a region of space via electromagnetic radiation.

This paper investigates the correlations between sensor signals in multipath environments created by correlated scatterers. We derive a closed form expression for the correlation in fields created by arbitrary scatterer correlations and scatterer powers, using Fourier techniques, and propose reasonable function forms for scatterer correlation. Simulations show notable differences from the uncorrelated scatterer case

This paper presents a method which allows to obtain expressions for the space-time cross correlation (STCC) and the space-frequency cross spectrum (SFCS) between signals at two receiver antennas on a mobile unit (MU) for an arbitrary scattering distribution surrounding the MU

We study the dimensions or degrees of freedom of random multipath fields in wireless communications. Random multipath fields are presented as solutions to the wave equation in an infinite-dimensional vector space. We prove a universal bound for the dimension of random multipath field in the mean square error sense. The derived maximum dimension is directly proportional to the radius of the two-dimensional spatial region where the field is coupled to. Using the Karhunen-Loeve expansion of multipath fields, we prove that, among all random multipath fields, isotropic random multipath achieves the maximum dimension bound. These results mathematically quantify the imprecise notion of rich scattering that is often used in multiple-antenna communication theory and show that even the richest scatterer (isotropic) has a finite intrinsic dimension

We propose a new broadband beamformer design technique which produces an optimal beampattern for any set of samples in space and time. The modal subspace decomposition (MSD) technique is based on projecting a desired pattern into the subspace of patterns achievable by a particular set of space-time sampling positions. This projection is the optimal achievable pattern, in the sense that it minimizes the mean-squared error (MSE) between the desired and actual patterns. The main advantage of the technique is versatility as it can produce optimal beamformers for both sparse and dense arrays, non-uniform and asynchronous time sampling, and dynamic arrays where sensors can move throughout space. It can also be applied to any beampattern type, including frequency-invariant and spot pattern design

In this paper, we derive an analytical expression for the exact pairwise error probability (PEP) of a differential space-time coded system operating over a spatially correlated slow fading channel. An analytic model for spatial correlation is used which fully accounts for antenna spacing, antenna geometry and non-isotropic scattering distributions. Inclusion of spatial information in error performance analysis provides valuable insights into the physical factors determining the performance of a differential space-time code (DSTC). Using this new PEP expression, we investigate the effects of antenna spacing, antenna geometries and azimuth power distribution parameters (angle of arrival/departure and angular spread) on the performance of a differential space-time block code (DSTBC) proposed in the literature for two transmit antennas.

Information transfer over a discrete time uncorrelated Rayleigh fading multiple input multiple output (MIMO) channel is considered, where neither the transmitter nor the receiver has the knowledge of the channel state information (CSI) except the fading statistics. We derive a capacity supremum with the receive antenna number at any signal to noise ratio (SNR) using Lagrange optimisation. We show that the asymptotic capacity is double logarithmic when the input power is large. We prove that to achieve the capacity, the amplitude of the multiple input needs to have a discrete distribution with a finite number of mass points, one of them necessarily located at the origin. We show how to compute the capacity numerically in multi-antenna configuration at any SNR with the discrete input using the Kuhn-Tucker condition for optimality. Furthermore, we show that the capacity with two mass points is optimal at low SNR signifying on-off keying. As the number of receive antennas increases, the maximum SNR at which two mass points are optimal decreases.

In this paper, we study the effect of directional random scattering on the capacity of multiple-input multiple-output (MIMO) systems. First, we use the spatial decomposition of the MIMO channel matrix to analyze the randomness (entropy) of directional scattering. The analysis shows that directional scatterers (with at least a null in the angular power spectrum) will no longer be random when the receiver observation radius is sufficiently large. Therefore, directional scattering limits the expected linear increase of MIMO capacity with increasing the number of antennas. Second, we consider the effect of receiver antenna arrangement (positions) on the capacity of MIMO systems. For any random scatterer with a given angular power spectrum, we show that it is possible to choose the receiver antenna arrangement with the optimum whitening of the MIMO channel matrix that, in turn, maximizes MIMO channel capacity.

In this paper, we use a novel MIMO channel model to characterize the dependence of ergodic capacity and diversity order on the joint statistics of the angular power density. The scattering environment of a MIMO channel is characterized by a double directional angular power distribution, describing the power transferred in each direction from transmitter aperture to receiver aperture. Angular power, which is typically separable Kronecker-modelled, is here generalized to include joint distribution properties using well-known bivariate probability density functions. We show that the joint properties of the power density, namely the shape and the orientation of power distribution contours, have significant impact on capacity and diversity of non-line-of-sight (NLOS) channels

In this paper, we study the effect of channel memory order on the information rate bounds in time-varying flat-fading (FF) channels. We model time variations of the FF channel with autoregressive (AR) processes with varying degrees of model order. We observe that in high SNR conditions (SNR 20 dB), the information rate penalty of not knowing the AR channel is a non-increasing function of the AR model order. This is expected, since the AR channel predictability cannot decrease with increasing its order. However, in low SNR conditions, the information rate penalty in low-order AR channels can be lower than those in high-order AR channels. Likewise, the intuitive and universal monotonic increase of the information rate bounds with the AR model order is only observed in almost noiseless conditions. In the low SNR regime, however, the achievable information rate bounds in low-order AR channels can be higher than those in high-order AR channels.

In this paper, we develop a channel simulator to generate the channel gains to an arbitrary array of receiver antennas, for a general class of non-line-of-sight channels. The channel scattering environment is defined by the angular power distribution as seen by the receiver. We derive the second order statistics of the channel gains in terms of the parameters of the angular power distribution. As an illustration of the channel simulator, we compare the performance of different direction-of-arrival techniques

This paper investigates the spatial-frequency channel characterization of ultra-wideband (UWB) wireless communication systems. First, a novel frequency dependent UWB channel model is constructed based on the theory of electromagnetic diffraction mechanism, which causes the field strength to vary with the frequency in each multipath. Then, we build a space-frequency model, which includes spatial characteristics such as angular power spectrum, and physical sampling points in space. The space-frequency model has two special cases (i) discrete multipath model, and (ii) cluster model, which can be readily used to generate channel data for any arbitrary set of sensor locations. The reconstruction results from channel measurements show the accurateness of the novel frequency dependent model, with reconstruction error decreasing by 40%, compared to the traditional Turin model

In this paper, we present a novel method of soundfield reproduction (SFR) for reverberant acoustic environments. Using an efficient parametrization of the acoustic transfer function (ATF) over a region of space, we devise a method for accurate SFR over the whole of the reproduction region. This method is based on a practical method of determining the ATF between each loudspeaker and the reproduction region.

Previous results have shown significant capacity gains by employing multiple antennas at both transmitter and receiver, however, due to physical size restraints (particularly at the receiver) these may not be obtained. In this paper, we consider the capacity behaviour of multi-antenna systems when the receiver sampling is constrained to a finite region of space. By characterizing the wavefield generated at the receiver due to transmitted signals and the scattering environment, a theoretically derived sampling threshold is shown to exist, at which the capacity growth is reduced from linear to logarithmic with increasing number of sampling outputs. Furthermore, this threshold is shown to be linearly dependent on the receiver region radius within which the sampling is constrained, and is independent of the sampling characteristics such as the antenna properties, array geometry, and/or array signal processing.

In this paper, performance and parameter optimization of RAKE reception for time-hopping ultra wideband (TH-UWB) systems is investigated when interchip interference (ICI) is taken into consideration. For a TH-UWB system, ICI is closely related to the relationship between the number of chips in a frame (N/sub c/), and the period of the TH code (N/sub f/). In a fixed data-rate case, larger N/sub f/ implies higher transmitted symbol signal-to-noise ratio (SNR) and larger ICI in RAKE fingers. So there is a tradeoff between N/sub c/ and N/sub f/ to optimize the RAKE performance. In this paper, two models are suggested to describe the ICI, and this tradeoff is investigated in the single user case based on the traditional Gaussian approximation method and other flexible methods. Given the lack of explicit knowledge of the relationship among N/sub f/, N/sub c/ and the interference, a rule of thumb is proposed to configure N/sub f/ and N/sub c/.

In this paper, the mutual information of a discrete time Rayleigh fading channel is considered, where neither the transmitter nor the receiver has the knowledge of the channel state information. We derive a closed form expression for the mutual information of single input single output Rayleigh fading channel when the input distribution is complex Gaussian. We compare the channel capacity and the mutual information attained with Gaussian input using the derived closed form expressions. Furthermore, we argue that the mutual information is bounded by the signal to noise ratio with the simulation results showing the sub optimality of Gaussian signalling in non-coherent Rayleigh fading channels.

We investigate the performance of acoustic equalization in reverberant environments. We first highlight an efficient general representation of a sound field using spherical harmonics. We then use this representation to develop a concise closed-form expression for robustness of equalization to sensor movement. This expression is used (i) to characterize equalization performance for a general class of non-isotropic sound fields and (ii) to quantify the improvements to equalizer robustness that can be obtained by using a directional microphone. This approach does not use any of the assumptions of statistical acoustics, but instead exploits the inherent properties of a sound field as described by the wave equation.

We propose a new method for range and bearing estimation of wideband sources based on a novel beamspace structure. The proposed structure consists of an orthogonal set of beamspace processors that have a frequency-invariant beampattern property, and can be steered easily to any nearfield range while maintaining the frequency-invariance of the spatial response. These properties mean that the proposed structure can perform coherent signal subspace processing for nearfield wideband sources, making it ideal for wideband range and bearing estimation. Performance of the estimator is demonstrated through Monte Carlo simulations.

We propose a new technique that facilitates the soft transition between a startup algorithm and a decision directed (DD) algorithm in blind adaptive equalizers. The algorithm-pair is combined using a reliability measure that is proportional to an estimate of the probability of the equalizer detecting a correct symbol. This measure takes into account both the equalized signal and its statistical distribution. The main feature of the technique is in the smooth and automatic switching from the startup algorithm to the DD algorithm and vice versa depending on the value of the reliability measure. This technique has been compared with the popular Benveniste-Goursat and stop-and-go algorithms and is shown to exhibit a faster rate of convergence and lower steady state error.

In this paper, we present the Cramer-Rao lower bounds (CKLBs) for the time delay estimation of UWB signals which could be tight lower bounds for the theoretical performance limits of UWB synchronizers. The CRLBs are investigated for both single pulse systems and time hopping systems in AWGN and multipath channels. Insights are given into the relationship between CRLBs for different Gaussian monocycles. It is found that larger number of multipath signals implies higher CRLBs and inferior performance of synchronizers, and multipath interference on CRLBs can not be eliminated completely except in very special cases. As every estimate of time delay could not be perfect, the least influence of the synchronization error on the performance of receivers is quantified.

Using a spatial region as an information bearing resource is considered and an analogy drawn with the work on essentially time- and band-limited signals by Slepian, Landau and Pollack. This paper considers the fundamental limits to the use of free-space as an information bearing resource, a concept for wireless communication which exploits space to achieve information transfer.

In this paper we present a new upper bound on the mutual information of MIMO systems. By characterizing the fundamental communication modes between two physical regions, we develop an intrinsic capacity which is independent of antenna array geometries and array signal processing, and depends only on the size of the regions and the statistics of the scattering environment.

We develop a general mathematical model for nearfield multipath scattering as a basis for studying the spatial limits imposed on multi-antenna wireless communication systems. This model generalizes the Herglotz wave function, which is an important tool in the study of inverse scattering problems, to a form where the scatterers can be nearfield. This permits the development of the most general form of spatial correlation which is known to constrain the capacity of wireless systems.

We investigate the behaviour of MIMO capacity when the size of the antenna array is constrained. By increasing the number of antennas within a small region in space the antenna array becomes dense and spatial correlation inhibits capacity growth. A theoretically derived antenna saturation point is shown to exist for dense array MIMO systems, at which there is no capacity growth with increasing antenna numbers. We show this saturation point increases linearly with the radius of the region containing the antenna array and is independent of the number of antennas.

A theoretically derived antenna saturation point is shown to exist for MIMO systems, at which the system suffers a capacity growth decrease from linear to logarithmic with increasing antenna numbers. We show this saturation point increases linearly with the radius of the region containing the receiver antennas and is independent of the number of antennas. Using an alternative formulation of capacity for MIMO systems we derive a closed form capacity expression, which uses the physics of signal propagation combined with statistics of the scattering environment. This expression gives the capacity of a MIMO system in terms of antenna placement and scattering environment and show that the saturation effect is due to spatial correlation between receiver antennas.

This paper investigates the performance of ultra-wideband (UWB) correlator receivers for Gaussian monocycles under the condition of equal mean power and provides constructive reference to the selection of pulses. Several channel situations are examined including ideal single user AWGN channel, non-ideal synchronous, multipath fading and multiple access interference. Both numerical and analytical techniques show that the shapes of pulses have notable impact on the performance of correlator receivers, especially on the interference resistance ability and signal-to-noise ratio (SNR) of the output. he results are also extended to the field of fractional bandwidth to better understand the possible relationship between fractional bandwidth and correlator receivers.

In this paper, some new results on the capacity of a typical M-ary pulse position modulation (M-PPM) time hopping (TH) ultra-wideband (UWB) systems are presented. Previous results using a "pure PPM model" are proven to exaggerate the real capacity of a UWB system. Based on an extended model containing correlator and soft decision decoding, the capacity is evaluated in the single-user case and in the case of a system with asynchronous multiple user interference (MUI) when the inputs are equiprobable. It is found that only when bit- signal-to-noise ratio (bit-SNR) is high enough, larger M leads to higher capacity; and for a specific M, the optimal values of PPM time offset parameter T/sub d/, which maximizes the capacity, are independent of bit-SNR. The influence of MUI on capacity is detrimental, especially in the case of high bit-SNR.

We derive an infinite series expression of the average signal-to-noise ratio (SNR) gain for a dual selection combining system under Nakagami-m fading using a series representation of the bivariate cumulative probability distribution function. The average SNR gain expression is shown to relate to the Nakagami shape factor m, the power correlation and the power allocation ratio.

A major problem in sound field reconstruction systems is how to record the higher order (> 1) harmonic components of a given sound field. Spherical harmonics analysis is used to establish theory and design of a higher order recording system, which comprises an array of small microphones arranged in a spherical configuration and associated signal processing. This result has implications to the advancement of future sound field reconstruction systems. An example of a third order system for operation over a 10∶1 frequency range of 340 Hz to 3.4 kHz is given.

We establish that an arbitrary narrowband multi path field in any circular region in two dimensional space has an intrinsic functional dimensionality of (πe) R/λ ≈ 8.54 R/λ. that scales only linearly with radius R/λ. in wavelengths. This result implies there is no such thing as an arbitrarily complicated multi path field. That is, a field generated by any number of nearfield and farfield, specular and diffuse multipath reflections is no more complicated than a field generated by a limited number plane waves. As such, there are limits on how rich multipath can be. This result has significant implications including means: i) to determine a parsimonious parameterization for arbitrary multipath fields, ii) of synthesizing arbitrary multi path fields with arbitrarily located nearfield or farfield, spatially discrete or continuous sources. We give examples of multipath field analysis and synthesis.

The well known results of the spatial correlation function for 2-dimensional and 3-dimensional diffuse fields of narrowband signals are generalised to the case of general distributions of scatterers. A method is presented which allows closed form expressions for the correlation function to be obtained for arbitrary scattering distribution functions. These closed form expressions are derived for a variety of commonly used scattering distribution functions.

The well-known results of the spatial correlation function for two-dimensional and three-dimensional diffuse fields of narrowband signals are generalized to the case of general distributions of scatterers. A method is presented that allows closed-form expressions for the correlation function to be obtained for arbitrary scattering distribution functions. These closed-form expressions are derived for a variety of commonly used scattering distribution functions.

Reproduction of a sound field is a fundamental problem in acoustic signal processing. In this paper, we use a spherical harmonics analysis to derive performance bounds on how well an array of loudspeakers can recreate a three-dimensional (3-D) plane-wave sound field within a spherical region of space. Specifically, we develop a relationship between the number of loudspeakers, the size of the reproduction sphere, the frequency range, and the desired accuracy. We also provide analogous results for the special case of reproduction of a two-dimensional (2-D) sound field. Results are verified through computer simulations.

An exact series representation for a nearfield spherically isotropic noise model is introduced. The proposed noise model can be utilized effectively to apply well-established farfield array processing algorithms for nearfield applications of sensor arrays. A simple array gain optimization is used to demonstrate the use of the new noise model.

This paper presents a new method of designing a beamformer having a desired broadband beampattern with focusing capability to operate at any radial distance from the array origin. An important consequence of our result is that the beamformer processing can be factored in to three levels of filtering: (i) beampattern independent elementary beamformers; (ii) beampattern shape dependent filters; and (iii) radial zooming filters where a single parameter can be adjusted to zoom-focus the array to a desired radial distance from the array origin. As an illustration the method is applied to the problem of producing a practical array design that achieves a frequency invariant beampattern over the frequency range of 1:10.

We establish the asymptotic equivalence, up to complex conjugation, of two problems: (i) determining the nearfield performance of a farfield beampattern specification, and (ii) determining the equivalent farfield beampattern corresponding to a nearfield beampattern specification. Using this reciprocity relationship we develop a computationally simple procedure to design a beamforming array to achieve a desired nearfield beampattern response. The superiority of this approach to existing methods, both in ease of design implementation and performance obtained, is illustrated by a design example.

This paper presents a method for nearfield frequency invariant broadband array beamforming. This means that we can design an array with associated signal processing to achieve a desired frequency invariant beam pattern (as a function of direction) at any nominal finite distance over an arbitrarily wide range of frequencies. The challenge here is to compensate accurately for the nearfield distortions (both magnitude and phase) which are more acute at the lower frequencies of the operating bandwidth than the higher frequencies. The methodology uses spherical harmonics to transform the nearfield broadband frequency invariant beampattern specification to an equivalent farfield frequency varying beampattern specification. A simulation example is presented to demonstrate the effectiveness of this method in producing a nearfield beam pattern which is frequency invariant over an octave bandwidth.