
Gongping Huang- Doctor of Philosophy
- Professor at Wuhan University
Gongping Huang
- Doctor of Philosophy
- Professor at Wuhan University
Professor of Electrical Engineering
About
88
Publications
16,629
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1,219
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Introduction
Gongping Huang is a Humboldt Research Fellow at the Chair of LMS in the University of Erlangen-Nuremberg, Germany. He was an Andrew and Erna Finci Viterbi postdoctoral research fellow in Technion - Israel Institute of Technology, Haifa, Israel. He received the BS and Ph.D. degree from the Center of Intelligent Acoustics and Immersive Communications (CIAIC), Northwestern Polytechnical University (NPU), Xian, China. He was a visiting researcher at INRS-EMT, University of Quebec, Montreal, Canada.
Current institution
Education
September 2008 - July 2012
Publications
Publications (88)
Deep learning based end-to-end multi-channel speech enhancement methods have achieved impressive performance by leveraging sub-band, cross-band, and spatial information. However, these methods often demand substantial computational resources, limiting their practicality on terminal devices. This paper presents a lightweight multi-channel speech enh...
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. The paper examines foundational developments in...
Modeling multitask relations in distributed networks has garnered considerable interest in recent years. In this paper, we present a novel rank-one model, where all the optimal vectors to be estimated are scaled versions of an unknown vector to be determined. By considering the rank-one relation, we develop a constrained centralized optimization pr...
Linear differential microphone arrays (LDMAs) are commonly integrated into thin and portable devices to achieve high-fidelity speech acquisition. Traditional LDMAs typically consist of only omnidirectional microphones, which impose limitations on their ability to produce steerable spatial responses due to constraints in array element directivity an...
Concentric circular microphone arrays have been used in a wide range of applications, such as teleconferencing systems and smarthome devices for speech signal acquisition. Such arrays are generally designed with omnidirectional sensors, and the associated beamformers are fully steerable but only in the sensors' plane. If operated in the three-dimen...
Nature is usually low rank! This means that there is nonnegligible redundancy in the observations. So, it is important to be able to translate this idea into equations in order to make things work better in practice. In this chapter, we discuss this concept and explain how it can be applied to beamforming. Then, we derive a large class of low-rank...
In this chapter, we study beamforming with very large arrays, i.e., arrays that contain a very large number of microphones. Conventional beamforming in this context, where a simple complex frequency-dependent weight is applied to each microphone, may not be very practical for obvious reasons such as high complexity, difficulty to accurately estimat...
Single microphone processing is a very popular technique in speech enhancement. It has been studied for several decades and has been implemented in many systems. This method has, obviously, great limitations. In this chapter, we explain why and show the importance of spatial information (with just two microphones) to fully understand why it leads t...
Distortionless beamforming plays a huge role in microphone array processing, in particular, and array processing, in general. Indeed, the most interesting and practical (fixed or adaptive) beamformers are distortionless. Therefore, it is of interest to study this important family of beamformers and understand how they really work. This is the objec...
In this chapter, we give a fresh perspective on binaural beamforming, where within the same process, we also try to take advantage of our binaural hearing system. Basically, after beamforming, we wish to place the desired speech signal and noise in different positions in the perceptual space in order to possibly improve intelligibility as compared...
Principal component analysis (PCA) is by far the most popular and useful dimensionality reduction technique that one can find in the literature. The objective of PCA is the reduction of the dimension of a random signal vector from M to P, where \(P \ll M\), with little loss of the useful information. It does so by preserving the variability of the...
Another fundamental way to perform beamforming is by considering pressure differences among microphones instead of direct pressure as in conventional beamforming. This leads to the co-called differential beamforming with at least two great advantages: frequency-invariant beampatterns and high directional gains. Although there are different strategi...
After a brief but insightful discussion on the limitations of single microphone processing and the unquestionable advantages of spatial information, in this chapter, we make a concise overview of the most fundamental concepts in microphone array processing. We start with the signal model by considering the general case of three-dimensional arrays....
In some applications, it may be more convenient to find a noise reference, which can then be used to adaptively cancel the additive noise at microphones. This chapter is concerned with this problem. Furthermore, as it will be explained, adaptive noise cancellation gives another insightful perspective of distortionless adaptive beamforming.KeywordsA...
This book explains the motivation for using microphone arrays as opposed to using a single sensor for sound acquisition. The book then goes on to summarize the most useful ideas, concepts, results, and new algorithms therein. The material presented in this work includes analysis of the advantages of using microphone arrays, including dimensionality...
Spatial information can help improve source separation performance. Numerous spatially informed source extraction methods based on the independent vector analysis (IVA) have been developed, which can achieve reasonably good performance in non-or weakly reverberant environments. However, the performance of those methods degrades quickly as the rever...
This paper studies the design of maximum directivity factor (MDF) beamformers based on uniform linear arrays (ULAs) consisting of acoustic vector sensors (AVSs). We first derive the main lobe constraints, which ensure that the beamformer's beampattern achieves a maximum in the look direction, and prove that any beamformer that satisfies the propose...
This paper is dedicated to the design of fully steerable linear differential microphone arrays (LDMAs). We analyze the steerable ideal spatial responses and explain why conventional LDMAs consisting of only omnidirectional microphones have limited steering ability. In order to circumvent this limitation, we suggest to use both omnidirectional and b...
Differential microphone arrays (DMAs) have demonstrated a great potential for high-fidelity acoustic and speech signal acquisition in a wide range of applications since such arrays are able to achieve frequency-invariant beampatterns with high directivity. Consequently, a great number of efforts have been devoted to the design of DMAs and the assoc...
Differential microphone arrays (DMAs) have demonstrated a great potential for solving the high-fidelity sound acquisition problem in a wide range of applications as they possess many good properties such as frequency-independent beampatterns with high directivity. A significant number of efforts have been devoted to the design of DMAs and the assoc...
A room Acoustic Impulse Response (RAIR), which represents the sound propagation channel via direct and reflection paths from a source position to a microphone, plays a leading role in a broad range of acoustic signal processing applications, e.g., echo cancellation. In practical acoustic environments, it is not uncommon that an RAIR may consist of...
Reverberation, whichis caused by late reflections, impairs not only speech quality but also intelligibility. Consequently, dereverberation, a process to mitigate the impact of reverberation, has attracted significant research interests. Numerous approaches have been developed in the literature, among which the weighted-prediction-error (WPE) one ha...
While differential beamforming with uniform linear arrays (ULAs) has been widely studied, there is little work so far regarding the design of differential beamformers with nonuniform linear arrays (NULAs). This article attempts to shed some light on the principles of differential beamforming with NULAs. We define spatial difference operators with N...
Differential beamforming, which measures the spatial derivatives of the acoustic pressure field, can be used in a wide range of small devices that require high-fidelity sound and speech acquisition as it can achieve frequency-invariant spatial responses with high directivity factors (DFs). Since a differential process is inherently sensitive to sen...
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...
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...
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,...
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...
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...
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...
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...
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...
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...
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...
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...
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...
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...
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...
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 beamforming with CMAs often suffers from the so-called null problem, which is caused by the zeros of the Bessel functions; then, concentric CMAs (CCMAs) ar...
This paper presents a theoretical study of differential beamforming with uniform linear arrays. By defining a forward spatial difference operator, any order of the spatial difference of the observed signals can be represented as a product of a difference operator matrix and the microphone array observations. Consequently, differential beamforming i...
In this paper, we study differential beamforming from a graph perspective. The microphone array used for differential beamforming is viewed as a graph, where its sensors correspond to the nodes, the number of microphones corresponds to the order of the graph, and linear spatial difference equations among microphones are related to graph edges. Spec...
Differential microphone arrays (DMAs) often encounter white noise amplification, especially at low frequencies. If the array geometry and the number of microphones are fixed, one can improve the white noise amplification problem by reducing the DMA order. With the existing differential beamforming methods, the DMA order can only be a positive integ...
Microphone array beamforming has been widely used in a wide range of acoustic applications. To make it effective in suppressing noise, yet being able to preserve the fidelity and quality of broadband speech signals of interest, the beamformer needs to be designed with high spatial gain, consistent responses at different frequencies , and high robus...
In microphone array beamforming, it is desirable to achieve a directive gain as high as possible for maximum acoustic noise rejection. The well-known superdirective beamformer was developed for this purpose; but it is sensitive to array imperfections such as sensors' self-noise, sensor placement errors, mismatch among sensor responses, etc, which r...
This paper is devoted to the study of the beamforming problem with circular microphone arrays (CMAs). It presents an approach to the design of beamformers with asymmetric and symmetric frequency-invariant beampatterns. We first discuss how to express a desired target directivity pattern, either symmetric or asymmetric, into a linear weighted combin...
Acoustic source localization (ASL) is a fundamental yet still challenging signal processing problem in sound acquisition, speech communication, and human-machine interfaces. Many ASL algorithms have been developed, such as the steered response power (SRP), the SRP-phase transform (SRP-PHAT), the minimum variance distortionless response (MVDR), the...
Reverberation is one of the most detrimental effects on the quality and performance of hands-free speech communication and human-machine interfaces. Among different approaches investigated in the literature, the weighted-prediction-error (WPE) method has exhibited some promising potential for dealing with reverberation. However , the WPE is highly...
This paper studies the problem of frequency-invariant beamforming with concentric circular microphone arrays (CCMAs) and presents an approach to the design of frequency-invariant and symmetric beampatterns. We first apply the Jacobi-Anger expansion to each ring of the CCMA to approximate the beampattern. The beamformer is then designed by using all...
This letter deals with the problem of differential beamforming with microphone arrays of arbitrary planar geometry. By approximating the beampattern with the Jacobi-Anger expansion, it develops an algorithm that can form any specified frequency-invariant beampattern with a microphone array of any planar geometry as long as the sensors' coordinates...
The maximum directivity (MD) beamformer with spherical microphone arrays has many salient features in processing broadband acoustic and speech signals while suppressing noise and reverberation; but it is sensitive to sensors' self-noise and mismatch among these sensors. One effective way to deal with this sensitivity is by increasing the number of...
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-invariant beamformers with either circular microphone ar...
A differential microphone array includes a plurality of microphones situated on a substantially planar platform, the plurality of microphones including a total number (M) of microphones and at least two subsets of the plurality of microphones situated along at least two substantially concentric ellipses with respect to a center, and a processing de...
Circular differential microphone arrays (CDMAs) have been extensively studied in speech and audio applications for their steering flexibility, potential to achieve frequency-invariant directivity patterns, and high directivity factors (DFs). However, CDMAs suffer from both white noise amplification and deep nulls in the DF and in the white noise ga...
This paper is dealing with two critical issues about uniform circular arrays (UCAs): frequency-invariant response and steering flexibility. It focuses on some optimal design of frequency-invariant beampatterns in any desired direction along the sensor plane. The major contributions are as follows. 1) We explain how to include the steering informati...
Superdirective beamforming has attracted a significant amount of research interest in speech and audio applications, since it can maximize the directivity factor (DF) given an array geometry and, therefore, is efficient in dealing with signal acquisition in diffuse-like noise environments. However, this beamformer is very sensitive to sensor self-n...
Superdirective beamformers with uniform circular arrays (UCAs) are often used in communication applications for their steering flexibility and potential high directivity factors (DFs). This paper proposes two classes of subspace superdirective beamformers. The first class is based on the diagonalization of the noise pseudo-coherence matrix with the...
This paper presents an approach to the direction-of-arrival (DOA) estimation problem in acoustic environments using microphone arrays. It works in the short-time Fourier transform (STFT) domain. It first transforms the noisy speech signals received at the array into the STFT domain. A Householder transformation is then constructed and applied to th...
Time delay estimation, which serves as the first stage that feeds into subsequent processors for localizing and tracking radiating sources, has been an active research topic for decades. This paper deals with time delay estimation in room acoustic environments. The major focus is on developing algorithms that can achieve robust time delay estimates...
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 of the optimal filtering matrix. The squared Pear...
This paper investigates a parametric gain approach to single-channel noise reduction in the frequency domain. In comparison with the tra ditional 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-squared error (MSE) between the noise and the noise...
Noise reduction is a problem of recovering a speech signal of interest from its noisy observations. Since the objective of the problem is to reduce noise, thereby improving the signal-to-noise ratio (SNR), it is natural to consider the use of maximum SNR filters. However, the maximum SNR filters, if not designed properly, may introduce significant...
This paper is devoted to the study and analysis of the maximum signal-to-noise ratio (SNR) filters for noise reduction both in the time and short-time Fourier transform (STFT) domains with one single microphone and multiple microphones. In the time domain, we show that the maximum SNR filters can significantly increase the SNR but at the expense of...
This paper studies the problem of single-channel noise reduction in the time domain. Based on some orthogonal decomposition developed recently and the squared Pearson correlation coefficient (SPCC), several noise reduction filters are derived. We will show that the optimization of the SPCC leads to the Wiener, minimum variance distortionless respon...
Questions
Question (1)
There are many techniques aims at improving the audio quality in communications system: echo cancellation, direction-of-arrive estimation, beamforming, noise reduction.
So, I want to ask, what is the processing sequence (namely, echo cancellation, direction-of-arrive estimation, beamforming, noise reduction) in a real communications system (like Video conference), and why?