Jacob Benesty

Jacob Benesty
Institut National de la Recherche Scientifique | INRS · Energy, Materials and Telecommunications Research Centre

PhD

About

731
Publications
132,476
Reads
How we measure 'reads'
A 'read' is counted each time someone views a publication summary (such as the title, abstract, and list of authors), clicks on a figure, or views or downloads the full-text. Learn more
19,464
Citations
Introduction
Jacob Benesty received the Ph.D. degree in control and signal processing from Orsay University, France, in April 1991. He worked at Telecom Paris University and then at Bell Laboratories, in Murray Hill, NJ, USA. In May 2003, he joined the University of Quebec, INRS-EMT, Montreal, Quebec, Canada, as a Professor. He is also an Adjunct Professor with Aalborg University, Denmark, and a Guest Professor with Northwestern Polytechnical University, Xi'an, China.
Additional affiliations
October 1995 - June 2003
Bell Labs
Position
  • MTS

Publications

Publications (731)
Article
In this paper, the identification problem of bilinear forms with the Wiener filter is addressed. The contribution is twofold. First, a different approach is introduced, by defining the bilinear term with respect to the impulse responses of a spatiotemporal model, in the context of multiple-input/singleoutput (MISO) systems. Second, two versions of...
Article
Full-text available
Linear system identification is a key problem in many important applications, among which echo cancellation is a very challenging one. Due to the long length impulse responses (i.e., echo paths) to be identified, there is always room (and needs) to improve the performance of the echo cancellers, especially in terms of complexity, convergence rate,...
Article
Differential beamformers with small-size microphone arrays are very attractive for audio and speech signal acquisition thanks to their high directivity and frequency-invariant spatial responses. However, such beamformers often suffer from significant white noise amplification at low frequencies, which makes their implementation in real-world system...
Article
Full-text available
The superdirective beamformer, while attractive for processing broadband acoustic signals, often suffers from the problem of white noise amplification. So, its application requires well-designed acoustic arrays with sensors of extremely low self-noise level, which is difficult if not impossible to attain. In this paper, a new binaural superdirectiv...
Article
Full-text available
In linear system identification problems, it is important to reveal and exploit any specific intrinsic characteristic of the impulse responses, in order to improve the overall performance, especially in terms of the accuracy and complexity of the solution. In this paper, we focus on the nearest Kronecker product decomposition of the impulse respons...
Article
Full-text available
The principal issue in acoustic echo cancellation (AEC) is to estimate the impulse response between the loudspeaker and microphone of a hands-free communication device. This application can be addressed as a system identification problem, which can be solved by using an adaptive filter. The most common one for AEC is the normalized least-mean-squar...
Chapter
Nonlinear systems have been studied for a long time and have applications in numerous research fields. However, there is currently no global solution for nonlinear system identification, and different used approaches depend on the type of nonlinearity. An interesting class of nonlinear systems, with a wide range of popular applications, is represen...
Article
Full-text available
The multilinear system framework allows for the exploitation of the system identification problem from different perspectives in the context of various applications, such as nonlinear acoustic echo cancellation, multi-party audio conferencing, and video conferencing, in which the system could be modeled through parallel or cascaded filters. In this...
Article
This paper presents a stochastic model of the least-mean-square for bilinear forms (LMS-BF) algorithm in which the bilinear term is defined with respect to the temporal and spatial impulse responses of a multiple-input/single-output (MISO) spatiotemporal system. Specifically, taking into account uncorrelated and correlated Gaussian input data, an a...
Article
Full-text available
Efficiently solving a system identification problem represents an important step in numerous important applications. In this framework, some of the most popular solutions rely on the Wiener filter, which is widely used in practice. Moreover, it also represents a benchmark for other related optimization problems. In this paper, new insights into the...
Article
Circular microphone arrays (CMAs) and concentric CMAs (CCMAs) have been used in a wide range of applications such as smartspeakers and teleconferencing systems because of their flexible steering ability. Although many efforts have been devoted to beamforming with CCMAs, most existing methods consider only the 2-dimensional (2D) case and assume that...
Article
Full-text available
Loudspeaker arrays with high directivity are desirable in many acoustic and sound applications to direct sounds into a desired region. One way of designing such arrays is through the differential operator to maximize the directivity factor. However, this method generally works for linear arrays with endfire steering direction and its usage to gener...
Article
Full-text available
System identification problems are always challenging to address in applications that involve long impulse responses, especially in the framework of multichannel systems. In this context, the main goal of this review paper is to promote some recent developments that exploit decomposition-based approaches to multiple-input/single-output (MISO) syste...
Article
Improving intelligibility of a speech signal of interest from its observations (with a single microphone) corrupted by additive noise has long been a challenging problem. Motivated by important findings achieved in the psychoacoustic field, we propose in this work a deep learning based method to render the noise and desired speech in the perceptual...
Conference Paper
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 enhancement algorithms for processing microphone...
Conference Paper
Full-text available
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^2 microphones into (M − 1)^2 subarrays with each subarray being a 2 × 2 square array formed by four adjacent microphones. Then, differential beamforming is performed with each subarr...
Article
Full-text available
The Kalman filter represents a very popular signal processing tool, with a wide range of applications within many fields. Following a Bayesian framework, the Kalman filter recursively provides an optimal estimate of a set of unknown variables based on a set of noisy observations. Therefore, it fits system identification problems very well. Neverthe...
Article
Full-text available
Microphone arrays combined with beamforming have been widely used to solve many important acoustic problems in a wide range of applications. Much effort has been devoted in the literature to microphone array beamforming, among which the Kronecker product beamforming method developed recently has demonstrated some interesting properties. Generally,...
Conference Paper
Full-text available
In this paper, we introduce a robust approach for rectangular differential beamforming. We present a 2-D multistage spatial mean operator which operates independently on the columns and rows of the observation signals of a uniform rectangular array (URA). The multistage approach enables high flexibility: two design parameters, Qc and Qr, set the nu...
Conference Paper
Full-text available
In this paper, we introduce an optimal quadratic Wiener beamformer for magnitude estimation of a desired signal. For simplicity, we focus on a two-microphone array and develop an iterative algorithm for magnitude estimation based on a quadratic multichannel noise reduction approach. We analyze two test cases, with uncorrelated and correlated noises...
Article
Spatial information is important for human perception of speech and sound signals. However, this information is often either distorted or completely neglected in noise reduction because it is challenging, to say the least, to achieve optimal noise reduction and accurate spatial information preservation at the same time. This paper studies the probl...
Article
In this paper, we propose a frequency-domain adaptive line enhancer (ALE) to reduce nonstationary harmonic noise, such as medical equipment beeps, from a noisy speech signal captured by a single microphone. The reduction of nonstationary noise is very challenging, with the tradeoff between noise reduction and speech distortion, often resulting with...
Article
Full-text available
In this paper, we present a generalized approach for differential microphone array (DMA) beamforming in the short-time Fourier transform (STFT) domain. We propose a multistage beamforming approach which considers a Kronecker product (KP) decomposition of the global beamformer into two independent sub-beamformers. We derive differential KP beamforme...
Article
Full-text available
Tensor-based signal processing methods are usually employed when dealing with multidimensional data and/or systems with a large parameter space. In this paper, we present a family of tensor-based adaptive filtering algorithms, which are suitable for high-dimension system identification problems. The basic idea is to exploit a decomposition-based ap...
Article
First-order differential microphone arrays (FOD- MAs), which combine a small-spacing uniform linear array and a first-order differential beamformer, have been used in a wide range of applications for sound and speech signal acquisition. However, traditional FODMAs are not steerable and their main lobe can only be at the endfire directions. To circu...
Article
This paper studies signal models for microphone array beamforming in the short-time-Fourier-transform (STFT) domain with long acoustic impulse responses. The major contributions are as follows. First, the signal modeling problem is investigated in the STFT domain and a general decomposition is proposed for the convolved source signal. Second, new i...
Article
Full-text available
In microphone array beamforming, a high directional gain is always desired for acoustic noise and reverberation suppression; as a result, the superdirective beamformer has been of great interest in many applications. However, this beamformer is well known to be very sensitive to array imperfections. While much effort has been made to improve its ro...
Article
In this paper, an adaptive algorithm is derived by considering that the beamforming vector can be decomposed as a Kronecker product of two smaller vectors. Such a decomposition leads to a joint optimization problem, which is then solved by using an alternating optimization strategy along with the steepest-descent method. The resulting algorithm, te...
Chapter
In this chapter, we briefly explain how graphs work and then show how beamforming with linear difference equations can be studied from this perspective.
Chapter
This chapter is a generalization of the two previous ones. We clearly show how to do beamforming with any order linear difference equations by first explaining the signal model. Then, performance measures are defined and useful fixed and adaptive beamformers are derived in this general context, which also includes conventional beamforming as a part...
Chapter
This chapter is dedicated to the study of beamforming with second-order linear difference equations.
Chapter
In this chapter, we briefly review the well-established conventional linear beamforming technique in the two-dimensional scenario. We start by explaining the signal model. Then, some very important performance measures in this context are defined. Finally, the most well-known (fixed and adaptive) conventional beamformers are derived.
Chapter
We all know from our own previous work and other works that pressure differences among microphones is another fundamental way to looking at beamforming, especially when sensors are close to each others. Conventional beamforming does not include this important information in its formulation and in an explicit way. Although, there are different manne...
Chapter
The concept of linear difference beamforming can be applied to the general filtering technique for speech enhancement. This is the purpose of this chapter. First, we explain the signal model and show how linear difference filtering works in noise reduction. Second, we derive the most important performance measures. Finally, we develop some examples...
Article
Differential microphone arrays (DMAs), which are responsive to the differential acoustic pressure fields, have been used in a wide range of applications related to audio and speech. The core part of a DMA is the so-called differential beamformer, which is generally designed by placing a number of nulls in its beampattern to attenuate noise from som...
Article
Full-text available
Differential microphone arrays (DMAs) have been used in a wide range of applications for high-fidelity acoustic signal acquisition and enhancement. In the design of differential beamformers, three of the widely used measures are the direc-tivity factor (DF), the front-to-back ratio (FBR), and the white noise gain (WNG). The former two have been use...
Article
Differential beamformers have demonstrated a great potential in forming frequency-invariant beampatterns and achieving high directivity factors. Most conventional approaches design differential beamformers in such a way that their beampatterns resemble a desired or target beampattern. In this paper, we show how to design differential beamformers by...
Article
Reverberation impairs not only the speech quality, but also intelligibility. The weighted-prediction-error (WPE) method, which estimates the late reverberation component based on a multichannel linear predictor, is by far one of the most effective algorithms for dereverberation. Generally, the WPE prediction filter in every short-time-Fourier-trans...
Article
Time difference of arrival (TDOA) estimation, which often serves as the fundamental step for a source localization or a beamforming system, has a significant practical importance in a wide spectrum of applications. To deal with reverberation, the TDOA estimation problem is often transformed into one of identifying the relative acoustic impulse resp...
Article
Differential microphone arrays (DMAs) can achieve high directivity and frequency-invariant spatial response with small apertures; they also have a great potential to be used in a wide spectrum of applications for high-fidelity sound acquisition. Although many efforts have been made to address the design of linear DMAs (LDMAs), most developed method...
Article
Full-text available
Humanoid robots require to use microphone arrays to acquire speech signals from the human communication partner while suppressing noise, reverberation, and interferences. Unlike many other applications, microphone arrays in humanoid robots have to face the restrictions in size and geometry. To address these challenges, this paper presents an approa...
Conference Paper
Full-text available
In this work we define and analyze the bilinear models which replace the conventional linear operation used in many building blocks of machine learning (ML). The main idea is to devise the ML algorithms which are adapted to the objects they treat. In the case of monochromatic images, we show that the bilinear operation exploits better the structure...
Article
Differential beamforming combined with microphone arrays can be used in a wide range of applications related to acoustic and speech signal acquisition and recovery. A practical and useful method for designing differential beamformers is the so-called null-constrained method, which was developed based on linear arrays and requires only the nulls' in...
Article
Full-text available
High-dimensional system identification problems can be efficiently addressed based on tensor decompositions and modelling. In this paper, we design a recursive least-squares (RLS) algorithm tailored for the identification of trilinear forms, namely RLS-TF. In our framework, the trilinear form is related to the decomposition of a third-order tensor...