52 accepted papers
Dereverberation has long been a crucial research topic in speech processing, aiming to alleviate the adverse effects of reverberation in voice communication and speech interaction systems. Among existing approaches, forward convolutional prediction (FCP) has recently attracted attention. It typicall…
Fixed beamforming is widely used in practice since it does not depend on the estimation of noise statistics and provides relatively stable performance. However, a single beamformer cannot adapt to varying acoustic conditions, which limits its interference suppression capability. To address this, ada…
Online blind source separation is essential for both speech communication and human-machine interaction. Among existing approaches, overdetermined independent vector analysis (OverIVA) delivers strong performance by exploiting the statistical independence of source signals and the orthogonality betw…
This paper reviews pioneering works in microphone array processing and multichannel speech enhancement, highlighting historical achievements, technological evolution, commercialization aspects, and key challenges. It provides valuable insights into the progression and future direction of these areas…
Direction-of-arrival (DOA) estimation is a key process in microphone array systems. The steered response power-based minimum variance distortionless response (SRP-MVDR) method performs very well in challenging acoustic environments but suffers from exponential complexity as the number of microphones…
Superdirective beamformers are highly effective at suppressing directional interference and diffuse noise, but their practical use is often constrained by the problem of white noise amplification. Robust superdirective beamforming methods typically address this by imposing a constraint on the white…
Superdirective beamformers, used with small microphone arrays, are highly attractive due to their high directivity and frequency-invariant beampatterns, making them well-suited for processing broadband acoustic and speech signals. However, these beamformers are very sensitive to array imperfections…
Differential microphone arrays (DMAs) have garnered significant attention in recent research and development due to their high directivity and frequency-invariant beampatterns. However, DMAs frequently encounter substantial white noise amplification, which limits their practical applications. This p…
Microphone array beamforming is widely used to extract desired speech signals from noisy environments. While most research in this area focuses on utilizing spatial information, less attention is given to the intrinsic physical mechanisms underlying microphone array observations. This paper aims to…
Kronecker product beamforming is an effective technique for designing beamformers with nonuniform linear arrays (NULAs). However, current techniques are restricted to NULAs with specific configurations, where the steering vector of the array is represented as a Kronecker product of steering vectors…
Accurate modeling and analysis of a radiator mounted on an infinite baffle are crucial for understanding its acoustic radiation characteristics. This paper investigates the radiation behavior of a convex dome-shaped radiator in such a condition, showing that, under the far-field approximation, the p…
This paper studies the problem of acoustic source localization in room environments. It presents an improved steered response power (SRP) approach with low-complexity and trade-off prewhitening. This method consists of two steps. In the first one, the linear predictor that is used to model the speec…
Thanks to their high directivity, compact size, and reliable performance, differential microphone arrays (DMAs) have attracted great interest from both industry and academia as they have demonstrated great potential to be used in a wide range of applications for high-fidelity speech acquisition. Nev…
Differential microphone arrays (DMAs) can measure both the acoustic pressure field and the differential acoustic pressure fields, which gives them great advantages in a wide range of applications for acoustic and speech signal acquisition. The core component of DMAs is the so-called differential bea…
This paper is devoted to the problem of noise covariance matrix (NCM) estimation. It proposes a time-frequency masking based approach. We first present an optimal mask function based on the mean-squared error criterion. To estimate this mask, we employ the recently developed directional gain method…
This paper investigates the problem of designing differential beam-formers with planar microphone arrays to achieve not only the desired target directivity pattern but also control the beamwidth (BW) or sidelobe level (SLL). We first discuss the target directivity patterns and express the Dolph-Cheb…
This paper proposes a frequency-domain recursive least-squares (RLS) adaptive filtering algorithm for identifying time-varying acoustic systems in noisy environments. The Kronecker product (KP) is employed to decompose the model filter of the acoustic channel impulse response into two sets of short…
This paper studies the problem of target speaker signal exaction and antiphasic rendering with an array of microphones in the scenarios where there are two active speakers. Based on the important findings achieved in the psychoacoustic field as well as our recent works on single-channel speech enhan…
Dereverberation, a process to mitigate or eliminate the reverberation effect, plays an important role in hands-free speech communication and human-machine interfaces. Tremendous efforts have been devoted to this problem and various methods have been developed over the last three decades. Those metho…
While multiframe noise reduction filters, e.g., the multiframe Wiener and minimum variance distortionless response (MVDR) ones, have demonstrated great potential to improve both the subband and full-band signal-to-noise ratios (SNRs) by exploiting explicitly the interframe speech correlation, the im…
In applications involving system identification problems, some characteristics of the impulse response of the system to be identified are usually exploited to design adaptive algorithms with improved performance. In this context, this paper focuses on the identification of systems that own intrinsic…
Null directions are important parameters for differential beamformers, which play an important role on the beamforming performance. In this paper, we investigate the performance of differential beamformers as a function of the null directions. We first derive the directivity factor (DF) as an explic…
This paper is devoted to the problem of adaptive beamforming with small-spaced microphone arrays. In this context, the Wiener filter is an optimal beamformer in the mean-squared error (MSE) sense. However, it requires good estimates of the covariance matrices of the speech signal of interest and noi…
While differential beamformers have been widely used in voice communication and human-machine speech interface systems to enhance speech signals of interest, how to design such beamformers that on the one hand can achieve the highest possible directivity factor (DF) and on the other hand are able to…
This paper studies the problem of designing square differential microphone arrays (SDMAs). It presents a multistage approach, which first divides an SDMA composed of M <sup xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">2</sup> microphones into (M − 1) <sup…
Microphone arrays have been used in wide range of applications for sound acquisition and signal enhancement, the performance of which depends not only on the processing algorithms but also on the array geometry. A large number of efforts have been devoted to the development of beamforming and signal…
To identify acoustic systems (which are low-rank in nature) in non-Gaussian and Gaussian noise, a robust recursive least M-estimate adaptive filtering algorithm is developed in this paper by applying the nearest Kronecker product to decompose the acoustic impulse response. Two M-estimators, i.e., th…
Differential beamformers with concentric circular microphone arrays (CCMAs) are desirable for use in various applications since they can form frequency-invariant spatial responses, have better beam steering flexibility than linear arrays, and suffer less with beampattern irregularity and white noise…
Frequency-invariant beamforming with circular microphone arrays (CMAs) has drawn a significant amount of attention for its steering flexibility and high directivity. However, frequency-invariant beam-forming with CMAs often suffers from the so-called null problem, which is caused by the zeros of the…
To identify acoustic systems in non-Gaussian and Gaussian noises, a robust frequency-domain recursive least M-estimate (FRLM) adaptive filtering algorithm is proposed. The cost function of the adaptive filter is defined by using a robust time-domain M-estimator, while its update equation is derived…
Differential microphone arrays (DMAs), a class of welldesigned small-size arrays combined with differential beamforming, are very useful for processing broadband acoustic, audio, and speech signals in a wide range of applications. However, most efforts in the literature so far have been devoted to l…
The recursive least-squares (RLS) adaptive filter is an appealing choice in system identification problems, mainly due to its fast convergence rate. However, this algorithm is computationally very complex, which may make it useless for the identification of high length impulse responses, like in ech…
This paper presents a method to design optimal linear differential microphone arrays (DMAs) by optimizing the array geometry. By constraining the DMA beamformer to achieve a given target value of the directivity factor (DF) with a specified target frequency-invariant beampattern while achieving also…
This paper proposes a method for the design of flexible Kronecker product beamformers based on the decomposition of the steering vector of a physical array as a Kronecker product of steering vectors of two smaller virtual arrays. With this decomposition, the global beamforming filter is designed by…
Small aperture circular microphone arrays (CMAs) have been widely used in many applications such as teleconferencing, smartspeakers, and robotics. A critical component of such arrays is the differential beamformer, which can achieve relatively high spatial gains with the same beampatterns at most fr…
In this paper, we present a single-channel smoothing-and-filtering technique for noise reduction in the time domain. Unlike traditional noise reduction methods, which directly apply a noise reduction filter to the noisy signal, the developed technique achieves noise reduction in two steps. It first…
In this paper, we develop the Kalman filter for the identification of bilinear forms. In this framework, the bilinear term is defined with respect to the impulse responses of a spatiotemporal model, which resembles a multiple-input/single-output system. Recently, the identification of such bilinear…
Beamforming using microphone arrays has been widely used for enhancing speech signals of interest and suppressing noise and interference in a wide range of applications. In order to make it work, beamforming generally assumes that the speech source of interest and the interference source are inciden…
This paper studies the problem of frequency-invariant beamforming with concentric circular microphone arrays (CCMAs). We develop a beamforming algorithm based on an optimal approximation of the beamformer's beampattern with the Jacobi-Anger expansion. In comparison with the existing frequency-invari…
This paper deals with the problem of single-channel noise reduction. Thanks to the eigenvalue decomposition, we arrange the eigenvalues of the speech correlation matrix in such a way that all the spectral mode signal-to-noise ratios (SNRs) of the noisy speech are ordered in a descending manner. By m…
In recent years, signal processing with ad hoc microphone arrays has attracted a lot of attention. Speech enhancement in noisy, interfered, and reverberant environments is one of the problems targeted by ad hoc microphone arrays. Most of the proposed solutions require knowledge of fingerprints, such…
Time delay estimation (TDE) plays an important role in localizing and tracking radiating acoustic sources. Although many efforts have been devoted to this problem in the literature, the robustness of TDE with respect to noise and reverberation remains a great challenge for practical systems. In this…
This paper presents an approach to the multichannel noise reduction problem. It first transforms the multichannel noisy speech signals into the frequency domain. A Householder transformation is then constructed, which converts the multichannel coefficients in each frequency bin into two components:…
In this paper, a blind algorithm is proposed for speech enhancement in multi-speaker scenarios, in which interference rejection is the main objective. Here, the ad hoc array is broken into microphone duples which are used to partition the array into local sub-arrays. The core algorithm takes advanta…
This paper develops a single-channel noise cancelation filter in the short-time Fourier transform (STFT) domain by combining the subspace method and the optimal filtering technique via joint diagonalization of the desired clean speech and noise signal correlation matrices. This filter is shown to be…
Noise and reverberation can significantly affect the performance of time delay estimation (TDE) in room acoustic environments. The multichannel cross-correlation coefficient (MCCC) algorithm, which extends the traditional cross-correlation method from two to multiple channels, can exploit the spatia…
Although they have been intensively studied and used in many applications due to their high directivity factor (DF), superdirective beamformers are sensitive to sensor noise and mismatch between sensors. This paper studies the problem of superdirective beamforming combined with the joint diagonaliza…
In this work, we consider enhancement of multichannel speech recordings. Linear filtering and subspace approaches have been considered previously for solving the problem. The current linear filtering methods, although many variants exist, have limited control of noise reduction and speech distortion…
This paper investigates a parametric gain approach to single-channel noise reduction in the frequency domain. In comparison with the traditional parametric Wiener gain, the major novelty of this presented approach is that the parametric gain is formulated to estimate the noise by using the mean-squa…
Directivity pattern or beampattern is an important performance measure in all fixed beamformers. Given a microphone array, how to design the beamforming filter so that the resulting directivity pattern is close to the desired one is a critical issue. In this paper, we study the design of such patter…
This paper studies the problem of single-channel noise reduction in the time domain, where an estimate of a vector of the desired clean speech is achieved by filtering a frame of the noisy signal with a rectangular filtering matrix. The core issue with this problem formulation is then the estimation…
Speech enhancement with distributed arrays has been met with various methods. On the one hand, data independent methods require information about the position of sensors, so they are not suitable for dynamic geometries. On the other hand, Wiener-based methods cannot assure a distortionless output. T…