ORCID Profile
0000-0001-8942-5328
Current Organisation
Curtin University
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Research Topics
In Research Link Australia (RLA), "Research Topics" refer to ANZSRC FOR and SEO codes. These topics are either sourced from ANZSRC FOR and SEO codes listed in researchers' related grants or generated by a large language model (LLM) based on their publications.
Top 5 Research Topics
ANZSRC Field of Research (FoR)
Communications Technologies | Electrical and Electronic Engineering | Signal Processing | Signal Processing | Broadband Network Technology | Wireless Communications | Speech Recognition | Medical Devices | Circuits and Systems | Modem Technology | Optimisation | Optical And Photonic Systems | Artificial Intelligence and Image Processing | Calculus of Variations, Systems Theory and Control Theory | Radio Communications And Broadcasting Not Elsewhere Classified | Acoustics and Noise Control (excl. Architectural Acoustics) | Pattern Recognition
ANZSRC Socio-Economic Objective (SEO)
Network transmission equipment | Telecommunications | Voice equipment | Mobile Data Networks and Services | Telemetry Equipment | Information processing services | Network switching equipment | Occupational health (excl. economic development aspects) | Expanding Knowledge in Technology | Expanding Knowledge in Engineering | Expanding Knowledge in the Mathematical Sciences |
Related Links
Publications
New Insights Into Optimal Acoustic Feedback Cancellation
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 09-2013
Compressed sensing based channel estimation and impulsive noise cancelation in underwater acoustic OFDM systems
Publisher: IEEE
Date: 11-2016
Mel-Spectrographic Mask Estimation for Missing Data Speech Recognition using Short-Time-Fourier-Transform Ratio Estimators
Publisher: IEEE
Date: 04-2007
Performance of an equaliser for trans-European trunked radio
Publisher: IEEE
Date: 2001
Non-uniform Optimal Subband Beamforming: An Evaluation on Real Acoustic Measurements
Publisher: IEEE
Date: 2008
A spatial filtering approach to robust adaptive beaming
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 1992
DOI: 10.1109/8.166535
Optimization and evaluation of sigmoid function with a priori SNR estimate for real-time speech enhancement
Publisher: Elsevier BV
Date: 02-2013
Digital Signal Processing for Hearing Instruments
Publisher: Springer Science and Business Media LLC
Date: 03-11-2009
DOI: 10.1155/2009/898576
Multi-Speaker Separation Employing Microphone Array and Vertex Finding Algorithm
Publisher: IEEE
Date: 09-2018
Acoustic Feedback Suppression for Multi-Microphone Hearing Devices Using a Soft-Constrained Null-Steering Beamformer
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 2020
Chebyshev optimization for the design of broadband beamformers in the near field
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 1998
DOI: 10.1109/82.659466
Robust Beamformer Design Against Mismatch in Microphone Characteristics and Acoustic Environment
Publisher: IEEE
Date: 09-2018
A hybrid iterative MIMO detection algorithm: Partial Gaussian approach with integer programming
Publisher: IEEE
Date: 10-2014
Towards the use of full covariance models for missing data speaker recognition
Publisher: IEEE
Date: 03-2008
A spectral slit approach to doubletalk detection
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 03-2012
Design of broadband beamformers with low complexity
Publisher: Springer Science and Business Media LLC
Date: 09-03-2012
DOI: 10.1186/1687-6180-2012-62
Abstract: In this article, we consider the design of broadband beamformers with low complexity. In fact, the design problem is multi-objective in nature, trading off between speech distortion and noise suppression. Finding a balance between these two objectives is important in order to achieve a desired sound quality. These measures are introduced as the objectives here. The design can then be obtained via a bi-objective integer programming problem, where the coefficients of the filters are expressed as sums of signed powers-of-two terms. We study two different integer spaces and penalty functions for solving the problem. Then, an algorithm based on a discrete filled function is developed for finding the optimal design. In order to illustrate the effectiveness of the algorithm, real data is used and two broadband beamformers are demonstrated.
Pilot-subcarrier based impulsive noise mitigation for underwater acoustic OFDM systems
Publisher: ACM Press
Date: 2016
Design of Robust Broadband Beamformers With Discrete Coefficients and Least Squared Criterion
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 12-2013
A hybrid method for the design of oversampled uniform DFT filter banks
Publisher: Elsevier BV
Date: 07-2006
Improved a priori snr estimation in speech enhancement
Publisher: IEEE
Date: 12-2017
Proportionate algorithms for two-microphone active feedback cancellation
Publisher: IEEE
Date: 08-2015
Real-Time Implementation of a Particle Filter with Integrated Voice Activity Detector for Acoustic Speaker Tracking
Publisher: IEEE
Date: 12-2006
Multicriteria Design of Oversampled Uniform DFT Filter Banks
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 06-2004
Reconfigurable FPGA-based switching path frequency-domain echo canceller with applications to voice control device
Publisher: Elsevier BV
Date: 03-2012
Design of robust steerable broadband beamformers incorporating microphone gain and phase error characteristics
Publisher: IEEE
Date: 05-2011
Design of Steerable Spherical Broadband Beamformers With Flexible Sensor Configurations
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 02-2013
Optimized decision delays in finite-length MIMO DFE
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 05-2005
A LabVIEW-based implementation of real-time underwater acoustic OFDM system
Publisher: IEEE
Date: 12-2017
Second-order-based blind signal separation in reverberant environments
Publisher: Informa UK Limited
Date: 06-12-2014
A robust transform domain echo canceller employing a parallel filter structure
Publisher: Elsevier BV
Date: 12-2006
Bayesian noise estimation in the modulation domain
Publisher: Elsevier BV
Date: 02-2018
Adaptive beamforming: Spatial filter designed blocking matrix
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 1994
DOI: 10.1109/48.338394
Non-uniform DFT filter banks design with semi-definite programming
Publisher: IEEE
Date: 2003
Null-steering beamformer for acoustic feedback cancellation in a multi-microphone earpiece optimizing the maximum stable gain
Publisher: IEEE
Date: 03-2017
Performance improvements for sector antennas using feature extraction and spatial interference cancellation
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 11-2002
A new design method for broadband microphone arrays for speech input in automobiles
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 07-2002
On the Indoor Beamformer Design With Reverberation
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 08-2014
Multichannel filters for speech recognition using a particle swarm optimization
Publisher: IEEE
Date: 12-2012
RUNX1cRegulates Hematopoietic Differentiation of Human Pluripotent Stem Cells Possibly in Cooperation with Proinflammatory Signaling
Publisher: Oxford University Press (OUP)
Date: 23-09-2017
DOI: 10.1002/STEM.2700
Abstract: Runt-related transcription factor 1 (Runx1) is a master hematopoietic transcription factor essential for hematopoietic stem cell (HSC) emergence. Runx1-deficient mice die during early embryogenesis due to the inability to establish definitive hematopoiesis. Here, we have used human pluripotent stem cells (hPSCs) as model to study the role of RUNX1 in human embryonic hematopoiesis. Although the three RUNX1 isoforms a, b, and c were induced in CD45+ hematopoietic cells, RUNX1c was the only isoform induced in hematoendothelial progenitors (HEPs)/hemogenic endothelium. Constitutive expression of RUNX1c in human embryonic stem cells enhanced the appearance of HEPs, including hemogenic (CD43+) HEPs and promoted subsequent differentiation into blood cells. Conversely, specific deletion of RUNX1c dramatically reduced the generation of hematopoietic cells from HEPs, indicating that RUNX1c is a master regulator of human hematopoietic development. Gene expression profiling of HEPs revealed a RUNX1c-induced proinflammatory molecular signature, supporting previous studies demonstrating proinflammatory signaling as a regulator of HSC emergence. Collectively, RUNX1c orchestrates hematopoietic specification of hPSCs, possibly in cooperation with proinflammatory signaling.
An improved soft decision based noise power estimation employing adaptive prior and conditional smoothing
Publisher: IEEE
Date: 09-2016
Quadratic programming for the design of two-dimensional weighted Chebyshev FIR filters with incomplete specifications
Publisher: Elsevier BV
Date: 12-1997
Uniform DFT Filter Bank with Finite Precision Prototype Filters
Publisher: IEEE
Date: 09-2018
Reverberation-Time Prediction Method for Room Impulse Responses Simulated with the Image-Source Model
Publisher: IEEE
Date: 10-2007
A Decision-Directed Adaptive Gain Equalizer for Assistive Hearing Instruments
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 08-2014
Post-Filtering Techniques For Directive Non-Stationary Source Combined With Stationary Noise Utilizing Spatial Spectral Processing
Publisher: IEEE
Date: 12-2006
Blind MIMO Equalization in the Frequency Domain
Publisher: IEEE
Date: 2005
Design of subsets of complex spreading sequences
Publisher: IEEE
Date: 2002
Modeling of Motion Dynamics and its Influence on the Performance of a Particle Filter for Acoustic Speaker Tracking
Publisher: IEEE
Date: 10-2007
Real time noise suppression in social settings comprising a mixture of non-stationary anc transient noise
Publisher: IEEE
Date: 08-2017
On a New SDP-SOCP Method for Acoustic Source Localization Problem
Publisher: Association for Computing Machinery (ACM)
Date: 25-10-2016
DOI: 10.1145/2968449
Abstract: Acoustic source localization has many important applications. Convex relaxation provides a viable approach of obtaining good estimates very efficiently. There are two popular convex relaxation methods using either semi-definite programming (SDP) or second-order cone programming (SOCP). However, the performances of the methods have not been studied properly in the literature and there is no comparison in terms of accuracy and performance. The aims of this article are twofold. First of all, we study and compare several convex relaxation methods. We demonstrate, by numerical ex les, that most of the convex relaxation methods cannot localize the source exactly, even in the performance limit when the time difference of arrival (TDOA) information is exact. In addressing this problem, we propose a novel mixed SDP-SOCP relaxation model and study the characteristics of the optimal solutions and its localizable region. Furthermore, an error correction scheme for the proposed SDP-SOCP model is developed so that exact localization can be achieved in the performance limit. Experimental data have been collected in a room with two different array configurations to demonstrate our proposed approach.
Subband generalized sidelobe canceller - a constrained region approach
Publisher: IEEE
Date: 2003
A Factor Graph Approach to Exploiting Cyclic Prefix for Equalization in OFDM Systems
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 12-2013
Acoustic feedback cancellation for a multi-microphone earpiece based on a null-steering beamformer
Publisher: IEEE
Date: 09-2016
A multi-DSP implementation of a broad-band adaptive beamformer for use in a hands-free mobile radio telephone
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 02-1991
DOI: 10.1109/25.69988
A Subband Space Constrained Beamformer incorporating Voice Activity Detection
Publisher: IEEE
Date: 2005
Microphone array speech processing
Publisher: Springer Science and Business Media LLC
Date: 16-09-2010
DOI: 10.1155/2010/694216
A Blind Approach to Joint Noise and Acoustic Echo Cancellation
Publisher: IEEE
Date: 2005
Noise estimation with lowcomplexity for speech enhancement
Publisher: IEEE
Date: 10-2011
Advances in Multimicrophone Speech Processing
Publisher: Springer Science and Business Media LLC
Date: 18-04-2006
On the integration of time-frequency masking speech separation and recognition in underdetermined environments
Publisher: IEEE
Date: 11-2012
Semi-infinite quadratic optimization method using dual parametrization with application to antenna array
Publisher: Informa UK Limited
Date: 02-2006
On the Selection of Significant Channel Parameters for OFDM Systems: A Hypotheses Testing Based Approach
Publisher: IEEE
Date: 08-2006
A combined variable step size strategy for two microphones acoustic feedback cancellation using proportionate algorithms
Publisher: IEEE
Date: 12-2017
Joint Channel Estimation and Impulsive Noise Mitigation in Underwater Acoustic OFDM Communication Systems
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 09-2017
Performance Analysis of Verification-Based Decoding for Packet-Based LDPC Codes over Binary Symmetric Channel
Publisher: IEEE
Date: 06-2009
Distributed Microphone Array Localization Problem via SDP-SOCP Method
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 2023
Enhanced Verification-Based Decoding for Packet-Based LDPC Codes
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 02-2008
A New Evidence Model for Missing Data Speech Recognition With Applications in Reverberant Multi-Source Environments
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 02-2011
Accelerated Conjugate Gradient for Second-Order Blind Signal Separation
Publisher: MDPI AG
Date: 11-11-2022
Abstract: This paper proposes a new adaptive algorithm for the second-order blind signal separation (BSS) problem with convolutive mixtures by utilising a combination of an accelerated gradient and a conjugate gradient method. For each iteration of the adaptive algorithm, the search point and the search direction are obtained based on the current and the previous iterations. The algorithm efficiently calculates the step size for the accelerated conjugate gradient algorithm in each iteration. Simulation results show that the proposed accelerated conjugate gradient algorithm with optimal step size converges faster than the accelerated descent algorithm and the steepest descent algorithm with optimal step size while having lower computational complexity. In particular, the number of iterations required for convergence of the accelerated conjugate gradient algorithm is significantly lower than the accelerated descent algorithm and the steepest descent algorithm. In addition, the proposed system achieves improvement in terms of the signal to interference ratio and signal to noise ratio for the dominant speech outputs.
Analysis and synthesis of signal set with constrained magnitude spectrum
Publisher: IEEE
Date: 2001
Comparison of pulse propagation and gain saturation characteristics among different input pulse shapes in semiconductor optical amplifiers
Publisher: Elsevier BV
Date: 2016
Weighted Chebyshev approximation for the design of broadband beamformers using quadratic programming
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 07-1994
DOI: 10.1109/97.311804
Utilizing neural network and critical band processing for speech enhancement
Publisher: IEEE
Date: 12-2017
A Fourier Based Method for Approximating the Joint Detection Probability in MIMO Communications
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 06-2012
Coherence Function and Adaptive Noise Cancellation Performance of an Acoustic Sensor System for Use in Detecting Coronary Artery Disease
Publisher: MDPI AG
Date: 31-08-2022
DOI: 10.3390/S22176591
Abstract: Adaptive noise cancellation is a useful linear technique to attenuate unwanted background noise that cannot be removed using traditional frequency-selective filters. Usually, this is due to the signal and noise co-existing in the same frequency band. This paper tests a weighted least mean squares (WLMS) algorithm on a stethoscope system for use in detecting coronary artery disease in the presence of background noise. Each stethoscope is equipped with two microphones: one used to detect heart signals and one used to detect background noise. The WLMS method was used for four different sources of background noise whilst measuring a heartbeat, including a single tone, multiple tones, hospital/clinic noise, and breathing noise. The magnitude-squared coherence between both microphones was unity for the tone scenarios, resulting in complete attenuation. For the other background noise sources, a less-than-unity magnitude-squared coherence resulted in minor and no attenuation. Thus, the coherence function is a tool that can be used to predict the amount of attenuation achievable by linear adaptive noise-cancellation techniques, such as WLMS, as presented in this article.
A low complexity iterative soft-decision feedback MMSE-PIC detection algorithm for massive MIMO
Publisher: IEEE
Date: 04-2015
A multi-filter system for speech enhancement under low signal-to-noise ratios
Publisher: American Institute of Mathematical Sciences (AIMS)
Date: 2009
A Bayesian Filter for Multi-View 3D Multi-Object Tracking With Occlusion Handling
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 05-2022
FPGA multi-filter system for speech enhancement via multi-criteria optimization
Publisher: Elsevier BV
Date: 08-2014
Performance limits of the broadband generalized sidelobe cancelling structure in an isotropic noise field
Publisher: Acoustical Society of America (ASA)
Date: 02-2000
DOI: 10.1121/1.428570
Abstract: In acoustic applications, the signal source of interest is generally broadband and the background noise field is predominantly isotropic. In this letter, the performance of the broadband generalized sidelobe canceller, optimized with respect to an isotropic noise field, is analyzed and compared against the performance of the conventional summing beamformer. The comparison is based on an ideal continuous-time infinite-length generalized sidelobe canceller with the aim of determining the performance limits of this beamforming structure.
Improving adaptive feedback cancellation in hearing aids using an affine combination of filters
Publisher: IEEE
Date: 03-2016
Assistive listening headsets for high noise environments: Protection and communication
Publisher: IEEE
Date: 04-2015
Time-frequency clustering with weighted and contextual information for convolutive blind source separation
Publisher: IEEE
Date: 06-2014
Iterative Method for the Design of DFT Filter Bank
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 11-2004
A robust approach to reverberant blind source separation in the presence of noise for arbitrarily arranged sensors
Publisher: IEEE
Date: 03-2012
Source number estimation in reverberant conditions via full-band weighted, adaptive fuzzy c-means clustering
Publisher: IEEE
Date: 05-2014
Speech Recognition Enhancement Using Beamforming and a Genetic Algorithm
Publisher: IEEE
Date: 2009
DOI: 10.1109/NSS.2009.44
Adaptive array noise suppression of handsfree speaker input in cars
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 1993
DOI: 10.1109/25.260760
Near-Field Broadband Beamformer Design via Multidimensional Semi-Infinite Linear Programming Techniques
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 11-2003
Null-Steering Beamformer-Based Feedback Cancellation for Multi-Microphone Hearing Aids With Incoming Signal Preservation
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 04-2019
Improved a priori SNR estimation with application in Log-MMSE speech estimation
Publisher: IEEE
Date: 10-2009
MMSE-Based Transceiver Design Algorithms for Interference MIMO Relay Systems
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 11-2015
Multichannel Signal Enhancement Algorithms for Assisted Listening Devices: Exploiting spatial diversity using multiple microphones
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 03-2015
Use of efficient frontier in microphone arrays
Publisher: Institution of Engineering and Technology (IET)
Date: 2006
DOI: 10.1049/EL:20061858
Post-Filtering with Multichannel Power Spectral Estimation Using Joint Diagonalization in Multi-Speaker Environments
Publisher: IEEE
Date: 08-2006
Improved practical variable step-size algorithm for adaptive feedback control in hearing aids
Publisher: IEEE
Date: 12-2016
Transceiver optimization for interference MIMO relay systems using the structure of relay matrix
Publisher: IEEE
Date: 10-2015
Robust Acoustic Direction of Arrival Estimation Using ROOT-SRP-PHAT, a Realtime Implementation
Publisher: IEEE
Date: 2005
Postfiltering Using Multichannel Spectral Estimation in Multispeaker Environments
Publisher: Springer Science and Business Media LLC
Date: 16-07-2007
DOI: 10.1155/2008/860360
Filter bank design for subband adaptive microphone arrays
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 2003
Design of oversampled uniform DFT filter banks with reduced inband aliasing and delay constraints
Publisher: IEEE
Date: 2001
Sound source localization for subband-based two speech separation in room environment
Publisher: IEEE
Date: 11-2201
Analysis of Two Microphone Method for Feedback Cancellation
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 2015
An adaptive low-complexity coherence-based beamformer
Publisher: IEEE
Date: 11-2013
Analysis of gain saturation characteristics in SOAs for different input pulse shapes
Publisher: IEEE
Date: 09-2014
A soft-in soft-out detection approach using partial Gaussian approximation
Publisher: IEEE
Date: 10-2012
An insight into the parametric multichannel wiener formulation
Publisher: IEEE
Date: 06-2011
Essential issues on solving optimal power flow problems using soft-computing
Publisher: American Institute of Mathematical Sciences (AIMS)
Date: 2014
Frequency Domain Improved Practical Variable Step-Size for Adaptive Feedback Cancellation Using Pre-Filters
Publisher: IEEE
Date: 09-2018
A Class of Pareto Optimal Binaural Beamformers
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 2022
Noise cancelling convergence rates for the LMS algorithm
Publisher: Elsevier BV
Date: 09-1991
Statistical Voice Activity Detection Using Low-Variance Spectrum Estimation and an Adaptive Threshold
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 03-2006
Joint Channel and Impulsive Noise Estimation in Underwater Acoustic OFDM Systems
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 11-2017
A low cost soft mapper for turbo equalization with high order modulation
Publisher: IEEE
Date: 11-2012
Design of digital waveform set for multiuser CDMA communications with matched filter receivers
Publisher: IEEE
Date: 2003
Closed-loop feedback cancellation utilizing two microphones and transform domain processing
Publisher: IEEE
Date: 05-2014
Blind Signal Separation Using Steepest Descent Method
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 08-2007
MMSE-based joint source and relay optimization for interference MIMO relay systems
Publisher: Springer Science and Business Media LLC
Date: 13-03-2015
Robust Source Localization in Reverberant Environments Based on Weighted Fuzzy Clustering
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 02-2009
Spreading Code Design Using a Global Optimization Method
Publisher: Springer Science and Business Media LLC
Date: 2005
Variational Bayesian Channel Tracking in High-speed Underwater Acoustic Communication
Publisher: ACM Press
Date: 2015
Orthogonal digital waveform set design for multiaccess communications systems
Publisher: IEEE
Date: 2002
Measurement driven birth model for the generalized labeled multi-Bernoulli filter
Publisher: IEEE
Date: 10-2016
Tracking and separation of multiple moving speech sources via cardinality balanced multi-target multi Bernoulli (CBMeMBer) filter and time frequency masking
Publisher: IEEE
Date: 10-2016
Stability-controlled hybrid adaptive feedback cancellation scheme for hearing aids
Publisher: Acoustical Society of America (ASA)
Date: 2018
DOI: 10.1121/1.5020269
Abstract: Adaptive feedback cancellation (AFC) techniques are common in modern hearing aid devices (HADs) since these techniques have been successful in increasing the stable gain. Accordingly, there has been a significant effort to improve AFC technology, especially for open-fitting and in-ear HADs, for which howling is more prevalent due to the large acoustic coupling between the loudspeaker and the microphone. In this paper, the authors propose a hybrid AFC (H-AFC) scheme that is able to shorten the time it takes to recover from howling. The proposed H-AFC scheme consists of a switched combination adaptive filter, which is controlled by a soft-clipping-based stability detector to select either the standard normalized least mean squares (NLMS) algorithm or the prediction-error-method (PEM) NLMS algorithm to update the adaptive filter. The standard NLMS algorithm is used to obtain fast convergence, while the PEM-NLMS algorithm is used to provide a low bias solution. This stability-controlled adaptation is hence the means to improve performance in terms of both convergence rate as well as misalignment, while only slightly increasing computational complexity. The proposed H-AFC scheme has been evaluated for both speech and music signals, resulting in a significantly improved convergence and re-convergence rate, i.e., a shorter howling period, as well as a lower average misalignment and a larger added stable gain compared to using either the NLMS or the PEM-NLMS algorithm alone. An objective evaluation using the perceptual evaluation of speech quality and the perceptual evaluation of audio quality measures shows that the proposed H-AFC scheme provides very high-quality speech and music signals. This has also been verified through a subjective listening experiment with N = 15 normal-hearing subjects using a multi-stimulus test with hidden reference and anchor, showing that the proposed H-AFC scheme results in a better perceptual quality than the state-of-the-art PEM-NLMS algorithm.
A Switched Algorithm for Adaptive Feedback Cancellation Using Pre-Filters in Hearing Aids
Publisher: MDPI AG
Date: 09-08-2021
DOI: 10.3390/AUDIOLRES11030037
Abstract: Acoustic coupling between microphone and loudspeaker is a significant problem in open-fit digital hearing aids. An open-fit compared to a close-fit hearing aid significantly lowers the signal quality and limits the achievable maximum stable gain. Adaptive feedback cancellation (AFC) enables an efficient approach to reduce the impact of acoustic coupling. However, without careful consideration, it can also introduce bias in estimating the feedback path due to the high correlation between the loudspeaker signal and the incoming signal, especially when the incoming signal is spectrally coloured, e.g., speech and music. The prediction error method (PEM) is well known for reducing this bias. The presented study aims to propose a switched PEM with soft-clipping (swPEMSC) that allows for further improvement in convergence/tracking rates, resulting in a better ability to recover from unstable/howling status. This swPEMSC employs a new update rule inspired by a soft-clipping based stability detector (SCSD). It allows to pick up either the PEMSC-NLMS or PEMSC-APA depending on the magnitude of the effective feedback signal howling corresponds to a large feedback signal. The PEMSC-NLMS with a small step-size ensures a low steady-state error, but slow convergence/tracking rates, while PEMSC-APA with a large step-size allows for fast convergence/tracking rates, but a high steady-state error. By combining those approaches, the proposed approach can take advantage of good characteristics from both. Experimental results using different types of incoming signals and an abrupt change of feedback paths show that the swPEMSC can shorten unstable periods (howling) by improving the convergence and tracking rates while retaining a low steady-state error and good signal quality.
Uniform FIR filterbank optimization with group delay specifications
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 11-2005
Combining null-steering and adaptive filtering for acoustic feedback cancellation in a multi-microphone earpiece
Publisher: IEEE
Date: 08-2017
Iterative Frequency Domain Equalization With Generalized Approximate Message Passing
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 06-2013
Soft-In Soft-Out Detection Using Partial Gaussian Approximation
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 2014
Simplified MMSE Precoding Design in Interference Two-Way MIMO Relay Systems
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 02-2016
Signal Processing Techniques for Assisted Listening [From the Guest Editors]
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 03-2015
Recognition of digital modulated signals based on statistical parameters
Publisher: IEEE
Date: 04-2010
Variable Digital Filter With Least-Square Criterion and Peak Gain Constraints
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 2007
An echo cancellation solution for voice control devices
Publisher: IEEE
Date: 05-2009
Enhanced data detection in OFDM systems using factor graph
Publisher: IEEE
Date: 10-2013
Maximum Likelihood Estimation and Cramer-Rao Lower Bounds for the Multichannel Spectral Evaluation in Hands-Free Communication
Publisher: IEEE
Date: 2005
On the design of signal sets with constrained magnitude spectrum and phase structure
Publisher: Wiley
Date: 2002
DOI: 10.1002/ACS.728
Investigation of Robust Features for Speech Recognition in Hostile Environments
Publisher: IEEE
Date: 2005
Enhancement of Speech Recognitions for Control Automation Using an Intelligent Particle Swarm Optimization
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 11-2012
A Combined On-Line PCA-ICA Algorithm for Blind Source Separation
Publisher: IEEE
Date: 2005
Non-causal delayless subband adaptive equalizer
Publisher: IEEE
Date: 05-2002
Filter Bank Design for DFT Based Transmultiplexers
Publisher: IEEE
Date: 2005
Microphone configuration for beamformer design using the Taguchi method
Publisher: Elsevier BV
Date: 2017
On the use of contextual time-frequency information for full-band clustering-based convolutive blind source separation
Publisher: IEEE
Date: 05-2014
Real-Time Adaptive Modulation Schemes for Underwater Acoustic OFDM Communication
Publisher: MDPI AG
Date: 30-04-2022
DOI: 10.3390/S22093436
Abstract: Adaptive modulation received significant attention for underwater acoustic (UA) communication systems with the aim of increasing the system efficiency. It is challenging to attain a high data rate in UA communication, as UA channels vary fast, along with the environmental factors. For a time-varying UA channel, a self-adaptive system is an attractive option, which can choose the best method according to the channel condition to guarantee the continuous connectivity and high performance constantly. A real-time orthogonal frequency- ision multiplexing (OFDM)-based adaptive UA communication system is presented in this paper, employing the National Instruments (NI) LabVIEW software and NI CompactDAQ device. In this paper, the received SNR is considered as a performance metric to select the transmission parameters, which are sent back to the transmitter for data transmission. In this research, a UA OFDM communication system is developed, employing adaptive modulation schemes for a nonstationary UA environment which allows to select subcarriers, modulation size, and allocate power adaptively to enhance the reliability of communication, guarantee continuous connectivity, and boost data rate. The recent UA communication experiments carried out in the Canning River, Western Australia, verify the performance of the proposed adaptive UA OFDM system, and the experimental results confirm the superiority of the proposed adaptive scheme.
A hybrid noise suppression filter for accuracy enhancement of commercial speech recognizers in varying noisy conditions
Publisher: Elsevier BV
Date: 2014
Enhanced Verification-Based Decoding for Packet-Based LDPC Codes over Wireless Channels
Publisher: IEEE
Date: 2008
Two-microphone acoustic feedback cancellation using NLMS based algorithms
Publisher: IEEE
Date: 06-2017
Blind signal separation using overcomplete subband representation
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 07-2001
DOI: 10.1109/89.928917
Speech enhancement strategy for speech recognition microcontroller under noisy environments
Publisher: Elsevier BV
Date: 10-2013
Affine projection algorithm for acoustic feedback cancellation using prediction error method in hearing aids
Publisher: IEEE
Date: 09-2016
FIR Variable Digital Filter With Signed Power-of-Two Coefficients
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 06-2007
Adaptive feedback control using improved variable step-size affine projection algorithm for hearing aids
Publisher: IEEE
Date: 12-2017
A Microphone Array Solution for Duplex Hands-Free Communication Systems
Publisher: IEEE
Date: 2007
Convolutive Blind Signal Separation With Post-Processing
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 09-2004
Optimal FIR subband beamforming for speech enhancement in multipath environments
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 11-2003
Adaptive Beamformer for Hands-Free Communication System in Noisy Environments
Publisher: IEEE
Date: 2005
Feedback Cancellation With Probe Shaping Compensation
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 03-2014
Evaluation of two-microphone acoustic feedback cancellation using uniform and non-uniform sub-bands in hearing aids
Publisher: IEEE
Date: 12-2015
Adaptive microphone array employing calibration signals: an analytical evaluation
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 05-1999
DOI: 10.1109/89.759030
Audio-Visual Based Online Multi-Source Separation
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 2022
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