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Noisy Vehicle Surveillance Camera: A System to Deter Noisy Vehicle in Smart City

A. Agha, R. Ranjan, W.S.Gan
Journal PaperSpecial issue on Acoustics in Smart Cities, Applied Acoustics, vol 117, Part B, 236–245, Feb 2017.

Abstract

With the increase in urbanization in modern cities, major roads and highways are built nearer to residential areas. Due to some inconsiderate driving behaviors, like accelerating and racing; illegal tailpipe modification; and aging vehicles, traffic noise is creating a nuisance to residents staying near highway, especially during late night or early morning. Unfortunately, these noisy vehicles often cannot be caught on the spot, unless traffic police is stationed at a certain place with their sound pressure level meter and camera. This latter approach is often not sustainable and incurs costly manpower resource. An automatic Noisy vehicle surveillance Camera (NoivelCam) system, which can be deployed in overhead structure across major highways near residential buildings, is being designed to capture noise level generated from passing vehicles and triggers a high-speed camera to capture the number plate of offending vehicles that exceed a certain noise level. Therefore, the proposed system provides time stamp, number plate, sound pressure level, audio, and video evidence of the offending vehicle and a means to issue an inspection notice to the owner of the vehicle. This paper provides a description of the NoivelCam, and some of its features to pinpoint noise generated from passing vehicle. An initial in-situ deployment of this system has been carried out in Singapore to test out its effectiveness in monitoring vehicle noise pressure level in a single lane on a highway.

Individualization of head-related transfer functions in the median plane using frontal projection headphones

K. Sunder W.S.Gan
Journal PaperJournal of Audio Engineering Society, vol 64, no. 12, pp. 1026–1041, Dec 2016.

Abstract

Using nonindividualized HRTFs in virtual audio synthesis produces front-back confusions, up-down reversals, in-head localization, and timbral coloration. Elevation and frontal localization are found to be most affected. In contrast, obtaining individualized HRTFs is a tedious process that involves complex acoustical measurements for each individual. Having a model of HRTF that does not involve tedious acoustical measurements would make the process much easier. In this research, individualization of the median plane HRTFs is explored using frontal projection headphones with a spherical head model because the frontal positioning of the headphone transducer inherently captures the idiosyncratic frontal spectral cues. To create the HRTFs, the important peaks (P1) and notches (N1, N2) are extracted first from the frontal headphone response and then shifted in frequency in accordance with the elevation angle. Detailed subjective experiments indicated that subjects were able to localize the virtual sound sources accurately with modeled HRTFs with results similar to individualized HRTFs.

Active Acoustic Window: Toward a Quieter Home

B. Lam, W.S.Gan
Journal Paper IEEE Potential, vol 35, no.1, pp 11-18, Jan/Feb 2016.

Abstract

Environmental noise (also known as noise pollution) is a prevalent feature of any urban soundscape. Of the numerous environmental noise sources (e.g., aircrafts, road traffic, railways, industries, and construction), the World Health Organization (WHO) has identified road traffic noise as one of the main contributors to urban noise pollution. Passive noise control (PNC) methods, where physical media are used to “shield” a listener from noise sources. Even though PNC methods are effective at damping noise over a large frequency range, they are less effective at the lower frequencies due to the thickness of media required. Therefore, active noise control (ANC) methods may hold the key to a practical noise mitigation solution for protecting the health of an ever-increasing urban population. ANC methods have been shown to be more spaceand cost-effective at attenuating low frequencies and are becoming increasingly realizable due to the recent development of efficient algorithms and powerful lowcost processors. Moreover, an ANC system retrofitted to open-windows may potentially attenuate low-frequency traffic noise while still allowing natural ventilation.

Generating Dual Beam from a Single steerable Parametric Loudspeaker

C. Shi, Y. Kajikawa, W.S.Gan
Journal PaperApplied Acoustics, Vol 99, pp 43-50, Dec 2015.

Abstract

The parametric loudspeaker utilizes an ultrasonic transducer array to transmit a directional sound beam in air based on the parametric array effect. In recent studies, phased array techniques have been applied to achieve controllable directivity patterns or to change the direction of the sound beam. Such a parametric loudspeaker is often referred to as a steerable parametric loudspeaker. In this paper, a dual beam generation method is elaborated. It aims to transmit two sound beams from just one steerable parametric loudspeaker. The two sound beams carries the same audio content to different locations. This dual beam generation method is compatible with the configuration of existing steerable parametric loudspeakers based on phased array techniques. As an algorithm solution, the dual beam generation method readily improves the flexibility of the steerable parametric loudspeaker.

Primary-Ambient Extraction Using Ambient Spectrum Estimation for Immersive Spatial Audio Reproduction

J.J. He, W.S.Gan, E.L. Tan
Journal PaperIEEE/ACM Transactions on Audio, Speech, and Language Processing, Vol.23, no.9, pp.1431,1444, Sept. 2015.

Abstract

The diversity of today's playback systems requires a flexible, efficient, and immersive reproduction of sound scenes in digital media. Spatial audio reproduction based on primary-ambient extraction (PAE) fulfills this objective, where accurate extraction of primary and ambient components from sound mixtures in channel-based audio is crucial. Severe extraction error was found in existing PAE approaches when dealing with sound mixtures that contain a relatively strong ambient component, a commonly encountered case in the sound scenes of digital media. In this paper, we propose a novel ambient spectrum estimation (ASE) framework to improve the performance of PAE. The ASE framework exploits the equal magnitude of the uncorrelated ambient components in two channels of a stereo signal, and reformulates the PAE problem into the problem of estimating either ambient phase or magnitude. In particular, we take advantage of the sparse characteristic of the primary components to derive sparse solutions for ASE based PAE, together with an approximate solution that can significantly reduce the computational cost. Our objective and subjective experimental results demonstrate that the proposed ASE approaches significantly outperform existing approaches, especially when the ambient component is relatively strong.

Time-Shifting Based Primary-Ambient Extraction for Spatial Audio Reproduction

J.J. He, W.S.Gan, E.L. Tan
Journal PaperIEEE/ACM Transactions on Audio, Speech, and Language Processing, vol.23, no.10, pp.1576,1588, Oct. 2015.

Abstract

One of the key issues in spatial audio analysis and reproduction is to decompose a signal into primary and ambient components based on their directional and diffuse spatial features, respectively. Existing approaches employed in primary-ambient extraction (PAE), such as principal component analysis (PCA), are mainly based on a basic stereo signal model. The performance of these PAE approaches has not been well studied for the input signals that do not satisfy all the assumptions of the stereo signal model. In practice, one such case commonly encountered is that the primary components of the stereo signal are partially correlated at zero lag, referred to as the primary-complex case. In this paper, we take PCA as a representative of existing PAE approaches and investigate the performance degradation of PAE with respect to the correlation of the primary components in the primary-complex case. A time-shifting technique is proposed in PAE to alleviate the performance degradation due to the low correlation of the primary components in such stereo signals. This technique involves time-shifting the input signal according to the estimated inter-channel time difference of the primary component prior to the signal decomposition using conventional PAE approaches. To avoid the switching artifacts caused by the varied time-shifting in successive time frames, overlapped output mapping is suggested. Based on the results from our experiments, PAE approaches with the proposed time-shifting technique are found to be superior to the conventional PAE approaches in terms of extraction accuracy and spatial accuracy.

Primary-Ambient Extraction Using Ambient Spectrum Estimation for Immersive Spatial Audio Reproduction

J.J. He, W.S.Gan, E.L. Tan
Journal PaperIEEE/ACM Transactions on Audio, Speech, and Language Processing, Vol.23, no.9, pp.1431,1444, Sept. 2015.

Abstract

The diversity of today's playback systems requires a flexible, efficient, and immersive reproduction of sound scenes in digital media. Spatial audio reproduction based on primary-ambient extraction (PAE) fulfills this objective, where accurate extraction of primary and ambient components from sound mixtures in channel-based audio is crucial. Severe extraction error was found in existing PAE approaches when dealing with sound mixtures that contain a relatively strong ambient component, a commonly encountered case in the sound scenes of digital media. In this paper, we propose a novel ambient spectrum estimation (ASE) framework to improve the performance of PAE. The ASE framework exploits the equal magnitude of the uncorrelated ambient components in two channels of a stereo signal, and reformulates the PAE problem into the problem of estimating either ambient phase or magnitude. In particular, we take advantage of the sparse characteristic of the primary components to derive sparse solutions for ASE based PAE, together with an approximate solution that can significantly reduce the computational cost. Our objective and subjective experimental results demonstrate that the proposed ASE approaches significantly outperform existing approaches, especially when the ambient component is relatively strong.

Primary-Ambient Extraction Using Ambient Phase Estimation with a Sparsity Constraint

J.J.He, W.S.Gan , E.L. Tan
Journal PaperIEEE Signal Processing Letter, Vol 22, No. 8, August 2015.

Abstract

Spatial audio reproduction addresses the growing commercial need to recreate an immersive listening experience of digital media content, such as movies and games. Primary-ambient extraction (PAE) is one of the key approaches to facilitate flexible and optimal rendering in spatial audio reproduction. Existing approaches, such as principal component analysis and time-frequency masking, often suffer from severe extraction error. This problem is more evident when the sound scene contains a relatively strong ambient component, which is frequently encountered in digital media. In this Letter, we propose a novel PAE approach by estimating the ambient phase with a sparsity constraint (APES). This approach exploits the equal magnitude of the uncorrelated ambient components in the two channels of a stereo signal and reformulates the PAE problem as an ambient phase estimation problem, which is then solved using the criterion that the primary component is sparse. Our experimental results demonstrate that the proposed approach significantly outperforms existing approaches, especially when the ambient component is relatively strong.

Modeling distance-dependent individual head-related transfer functions in the horizontal plane using frontal projection headphones

K Sunder, W.S.Gan, E.L. Tan
Journal PaperJournal of Acoustical Society of American, Vol 138, No. 1, pp 150-171, July 2015.

Abstract

The veracity of virtual audio is degraded by the use of non-individualized head-related transfer functions (HRTFs) due to the introduction of front-back, elevation confusions, and timbral coloration. Hence, an accurate reproduction of spatial sound demands the use of individualized HRTFs. Measuring distance-dependent individualized HRTFs can be extremely tedious, since it requires precise measurements at several distances in the proximal region (less than 1 m) for each individual. This paper proposes a technique to model distance-dependent individualized HRTFs in the horizontal plane using “frontal projection headphones playback” that does not require individualized measurements. The frontal projection headphones [Sunder, Tan, and Gan (2013). J. Audio Eng. Soc. 61, 989–1000] project the sound directly onto the pinnae from the front, and thus inherently create listener idiosyncratic pinna cues at the eardrum. Perceptual experiments were conducted to investigate cues (auditory parallax and interaural level differences) that aid distance perception in anechoic conditions. Interaural level differences were identified as the prominent cue for distance perception and a spherical head model was used to model these distance-dependent features. Detailed psychophysical experiments revealed that the modeled distance-dependent individualized HRTFs exhibited localization performance close to the measured distance-dependent individualized HRTFs for all subjects.

An Objective Analysis Method for Perceptual Quality of a Virtual Bass System

H Mu, W.S.Gan, E.L. Tan
Journal PaperAccepted for publication in the IEEE/ACM Transaction on Audio, Speech, and Language Processing, Vol 23, No. 5, pp 840-850, May 2015.

Abstract

Due to the physical size and frequency response constraints of miniaturized and flat panel loudspeakers, low frequency reproduction from these loudspeakers is generally limited and unsatisfactory. The virtual bass system (VBS) enhances the bass performance of such loudspeakers by tricking the human brain to perceive the fundamental frequency from its higher harmonics. Problematically, the additional harmonics from VBS can also introduce perceptual distortion and reduce the audio quality. Therefore, a reliable method to assess the quality of VBS-enhanced signals is necessary in the designing of VBS. Since subjective experiments are often time-consuming and may be inconsistent, it is desirable to develop an objective assessment method for VBS. Earlier studies only utilized some simple objective metrics, which generally do not consider the human auditory model and are unable to accurately predict the perceptual quality of VBS. In this paper, we introduce a perceptual quality-assessment method for VBS based on the model output variables (MOVs) of the ITU Recommendation ITU-R BS.1387. Suitable combinations of MOVs are selected to derive perceptual quality metrics that correlate closely to the subjective quality. A verification experiment is presented to justify the accuracy of the metrics.

Natural Sound Rendering for Headphones: Integration of Signal Processing Techniques

K Sunder, J.J.He, W.S.Gan, E.L. Tan
Journal PaperIEEE Signal Processing Magazine, Vol 32, No. 2, pp 100-113, March 2015.

Abstract

With the strong growth of assistive and personal listening devices, natural sound rendering over headphones is becoming a necessity for prolonged listening in multimedia and virtual reality applications. The aim of natural sound rendering is to naturally recreate the sound scenes with the spatial and timbral quality as natural as possible, so as to achieve a truly immersive listening experience. However, rendering natural sound over headphones encounters many challenges. This tutorial article presents signal processing techniques to tackle these challenges to assist human listening.

An overview of directivity control methods of the parametric array loudspeaker

C Shi, W.S.Gan
Journal PaperAPSIPA Transaction on Signal and Information Processing, pp 1 - 30 DOI: 10.1017/ATSIP.2014.18, Published online: 22 December 2014.

Abstract

A sound reproduction system usually consists of several types of loudspeakers to cater to sophisticated applications. The directivity of a loudspeaker is a measure of its efficiency in sending sounds to a particular direction instead of all directions. Demand to control the directivity of a sound reproduction system is gaining momentum with many new designs of directional loudspeakers, including the acoustic dome, horn loudspeaker, loudspeaker array, and parametric array loudspeaker (PAL). The PAL is an application of the parametric acoustic array in air, which generates a sound beam from the interaction of ultrasonic beams. The PAL has several desired features, such as its narrow beamwidth over a wide frequency range, low sound attenuation over a long distance, and ability to reproduce perceptually near sound images. The PAL is also advantageous in using a smaller driving unit to transmit an equally narrow sound beam as compared to the conventional loudspeaker and broadside loudspeaker array. An overview of directivity control methods of the PAL is presented in this paper. In particular, acoustic models and signal processing techniques in controlling the directivity of the PAL are emphasized.

Convergence Analysis of Narrowband Feedback Active Noise Control System with Imperfect Secondary-Path Estimation

L Wang, W.S.Gan, Sen M Kuo
Journal Paper IEEE Transaction on Audio, Speech, and Language Processing, Vol. 21, No.11, pp2403-2411,2013.

Abstract

In many practical active noise control (ANC) applications, feedback structure using estimated secondary path to synthesize reference signal is preferred under various conditions. This paper analyzes the convergence behavior of the narrowband feedback ANC systems with imperfect secondary path estimation. Existing approaches do not include the analysis of the reference signal synthesis errors due to its interrelated feedback nature. In this paper, the reconstruction error is modeled using the secondary path estimation error. Using this model, the effects of estimation errors on the convergence of the feedback ANC system is investigated. To further examine the effects of error in the filtered- x and filtered- y signal paths, these two paths are analyze separately to isolate the effects caused by these paths. Computer simulations are conducted to verify the theoretical analysis presented in the paper.

Recent Advances on Active Noise Control: Open Issues and Innovative Applications,” in the

Y Kajikawa, W.S.Gan , S.M. Kuo
Journal Paper Inauguration issue of APSIPA Transaction on Signal and Information Processing (Invited paper), 2012.

Abstract

The problem of acoustic noise is becoming increasingly serious with the growing use of industrial and medical equipment, appliances, and consumer electronics. Active noise control (ANC), based on the principle of superposition, was developed in the early 20th century to help reduce noise. However, ANC is still not widely used owing to the effectiveness of control algorithms, and to the physical and economical constraints of practical applications. In this paper, we briefly introduce some fundamental ANC algorithms and theoretical analyses, and focus on recent advances on signal processing algorithms, implementation techniques, challenges for innovative applications, and open issues for further research and development of ANC systems.

Embedded Signal Processing with the Micro Signal Architecture

W.S.Gan, Sen M. Kuo
Book Wiley-IEEE press, 2007 ISBN:0471738417
image

Description

This is a real-time digital signal processing textbook using the latest embedded Blackfin processor Analog Devices, Inc (ADI). 20% of the text is dedicated to general real-time signal processing principles. The remaining text provides an overview of the Blackfin processor, its programming, applications, and hands-on exercises for users. With all the practical examples given to expedite the learning development of Blackfin processors, the textbook doubles as a ready-to-use user's guide.

The book is based on a step-by-step approach in which readers are first introduced to the DSP systems and concepts. Although, basic DSP concepts are introduced to allow easy referencing, readers are recommended to complete a basic course on "Signals and Systems" before attempting to use this book. This is also the first textbook that illustrates graphical programming for embedded processor using the latest LabVIEW Embedded Module for the ADI Blackfin Processors. A solutions manual is available for adopters of the book from the Wiley editorial department.

Contents.

  • Preface
  • Acknowledgments
  • About the Authors
  • Chapter 1. Introduction
  • Chapter 2. Time-Domain Signals and Systems
  • Chapter 3. Frequency-Domain Analysis and Processing
  • Chapter 4. Digital Filtering
  • Chapter 5. Introduction to the Blackfin Processor
  • Chapter 6. Real-Time DSP Fundamentals and Implementation Considerations
  • Chapter 7. Memory System and Data Transfer
  • Chapter 8. Code Optimization and Power Management
  • Chapter 9. Practical DSP Applications: Audio Coding and Audio Effects
  • Chapter 10. Practical DSP Applications: Digital Image Processing
  • Appendix A. An Introduction to Graphical Programming with LabVIEW
  • Appendix B. Useful Websites
  • Appendix C. List of Files Used in Hands-On Experiments and Exercises
  • Appendix D. Updates of Experiments Using Visual DSP++ V4.5.
  • References
  • Index

Digital Signal Processors: Architectures, Implementations, and Applications

Sen M. Kuo, W.S.Gant
Book Prentice Hall, Published 03/26/2004 ISBN-10: 0130352144 • ISBN-13: 9780130352149
image

Description

This text offers students a hands-on approach to understanding architecture and programming of DSP processors, and the design of real-time DSP systems. It contains real-world applications, and implementation of DSP algorithms using both the fixed-point and floating-point processors.

Contents.

  • 1. Introduction to DSP Systems.
  • 2. Fundamentals of Digital Signal Processing.
  • 3. Implementation Considerations.
  • 4. Fixed-Point DSP Processors.
  • 5. Floating-Point DSP Processors.
  • 6. FIR Filtering.
  • 7. IIR Filtering.
  • 8. Fast Fourier Transforms.
  • 9. Adaptive Filtering.
  • Appendix A. Introduction to MATLAB and Simulink.
  • Appendix B. Additional Hands-on Experiments and Applications
  • Appendix C. Description of Files in Companion CD and Website.
  • Index.

Wave Field Synthesis: The Future of Spatial Audio

R Ranjan,W.S.Gan
Journal Paper IEEE Potentials, Vol. 32, No.2, pp 17-23, March/April 2013.

Abstract

We all are used to perceiving sound in a three-dimensional (3-D) world. In order to reproduce real-world sound in an enclosed room or theater, extensive study on how spatial sound can be created has been an active research topic for decades. Spatial audio is an illusion of creating sound objects that can be spatially positioned in a 3-D space by passing original sound tracks through a sound-rendering system and reproduced through multiple transducers, which are distributed around the listening space. The reproduced sound field aims to achieve a perception of spaciousness and sense of directivity of the sound objects. Ideally, such a sound reproduction system should give listeners a sense of an immersive 3-D sound experience. Spatial audio can primarily be divided into three types of sound reproduction techniques, namely, loudspeaker stereophony, binaural technology, and reconstruction using synthesis of the natural wave field [which includes Ambisonics and wave field synthesis (WFS)], as shown in Fig. 1(a).

Individualization of Binaural Synthesis Using Frontal Projection Headphones

K Sunder, E.L. Tan,W.S.Gan
Journal Paper IJournal of Audio Engineering Society, Vol 61, No. 12, pp989-1000, 2013.

Abstract

Non-individualized head related transfer functions (HRTF) limit the spatial accuracy of conventional side projection headphones. This research explores the use of a frontal projection headphone, which customizes the HRTF by introducing idiosyncratic pinna cues. In addition, a robust headphone equalization technique is recommended for frontal projection headphone playback to preserve the embedded personal pinna cues. Perceptual experiments validated the effectiveness of frontal headphone playback over the conventional headphones with reduced front-back confusions and improved frontal localization. It was also observed that the individual spectral cues created by the frontal projection are sufficient for front-back discrimination even with the high frequency pinna cues removed from the non-individual HRTF.

Fixed-point Square Roots Using L-bit Truncation

A Seth, W.S.Gan
Journal Paper IEEE Signal Processing Magazine, Vol. 28, No. 6, pp 149-153, Nov 2011.

Abstract

Square root (SQRT) is a common arithmetic operation used in many DSP algorithms. In this paper, we evaluate square rooting methods suitable for implementation on fixed-point (FxP) DSP processors with a fast multiplying unit. The finite wordlength effect on the square rooting methods is highlighted, and it is shown that the theoretically derived convergence rate for the Newton-Raphson (NR) based square rooting methods are not suitable for FxP processor. Also, the most efficient methods for 8-bit and 16-bit FxP processors are identified.

Arbitrary Resizing of Images in the Discrete Cosine Transform Domain

E. L. Tan, W.S.Gan, S Mitra
Journal PaperIET Image Processing journal, Vol. 5, No. 1, pp 73-86, 2011.

Abstract

This study presents a method to resize images by a rational factor of P/Q in the discrete cosine transform (DCT) domain, where P and Q are relatively prime integers larger than 1. Our method extends on the prior work of Mukherjee and Mitra, which utilises the spatial relationship of DCT coefficients between a block and its sub-blocks. To resize images by a factor of P/Q, the images are first up-sampled by a factor of P and then down-sampled by a factor of Q. Although this method produces resized images with good visual quality, it requires high computational cost. In this study, the authors generalise an observation found in the spatial relationship of the DCT coefficients between a block and its sub-blocks. Subsequently, a sparse matrix representation is derived from this observation to reduce the computational cost of the proposed method. To further reduce computational cost of the proposed method, a subset of up-sampled DCT coefficients is used in the down-sampling operation. From various experiments, the authors have determined the lowest number of up-sampled DCT coefficients to be used in the down-sampling operation without affecting the visual quality of the resized images. As compared to existing methods, the proposed method requires lower computational cost and produces resized images of good visual quality.

Perceptually Tuned Subband Coder for JPEG Journal of Real-Time Image Processing

E. L. Tan,W.S.Gan
Journal Paper Journal of Real-Time Image Processing, pp1-15, 16 Jan 2010.

Abstract

This paper presents a perceptually tuned subband image coder, which selectively operates on the subbands of an image to achieve good compression ratio. The proposed encoder divides an image into 8 × 8 sub-images and subsequently decomposes each sub-image into four subbands. A just noticeable distortion (JND) model is employed to remove all the perceptually insignificant subbands from the sub-images. The remaining subbands are processed by the subband discrete cosine transform (SBDCT). This reduces the computational cost of the SBDCT without introducing perceptible distortion. To further reduce the perceptual redundancy in the image, an image-dependent quantization matrix is computed from the JND profile of the image. Our simulation results indicate that our proposed encoder outperforms the JPEG baseline coder at bit rates from 0.25 to 0.9 bpp.

Efficient Algorithm and Architecture of Critical-Band Transform for Low-Power Speech Applications

C Wang, W.S.Gan
Journal Paper EURASIP Journal on Advances in Signal Processing (2007), Vol. 2007, Article ID 89264 , 2007

Abstract

An efficient algorithm and its corresponding VLSI architecture for the critical-band transform (CBT) are developed to approximate the critical-band filtering of the human ear. The CBT consists of a constant-bandwidth transform in the lower frequency range and a Brown constant- transform (CQT) in the higher frequency range. The corresponding VLSI architecture is proposed to achieve significant power efficiency by reducing the computational complexity, using pipeline and parallel processing, and applying the supply voltage scaling technique. A 21-band Bark scale CBT processor with a sampling rate of 16 kHz is designed and simulated. Simulation results verify its suitability for performing short-time spectral analysis on speech. It has a better fitting on the human ear critical-band analysis, significantly fewer computations, and therefore is more energy-efficient than other methods. With a 0.35 m CMOS technology, it calculates a 160-point speech in 4.99 milliseconds at 234 kHz. The power dissipation is 15.6 W at 1.1 V. It achieves 82.1 power reduction as compared to a benchmark 256-point FFT processor.

Efficient VLSI Architecture for Lifting-Based Discrete Wavelet Packet Transform

C Wang, W.S.Gan
Journal Paper IEEE Transaction on Circuit & System, Part II, Vol. 54, No. 5, pp 422-426, May 2007.

Abstract

This brief presents a novel very large-scale integration (VLSI) architecture for discrete wavelet packet transform (DWPT). By exploiting the in-place nature of the DWPT algorithm, this architecture has an efficient pipeline structure to implement high-throughput processing without any on-chip memory/first-in first out access. A folded architecture for lifting-based wavelet filters is proposed to compute the wavelet butterflies in different groups simultaneously at each decomposition level. According to the comparison results, the proposed VLSI architecture is more efficient than the previous proposed architectures in terms of memory access, hardware regularity and simplicity, and throughput. The folded architecture not only achieves a significant reduction in hardware cost but also maintains both the hardware utilization and high-throughput processing with comparison to the direct mapped tree-structured architecture.

Teaching DSP Software Development: From Design to Fixed-Point Implementations

W.S.Gan,Sen M Kuo
Journal Paper IEEE Transaction on Education, Vol. 49, No. 1, pp 122-131, Feb 2005.

Abstract

In this paper, a digital signal processing (DSP) software development process is described. It starts from the conceptual algorithm design and computer simulation using MATLAB, Simulink, or floating-point C programs. The finite-word-length analysis using MATLAB fixed-point functions or Simulink follows with fixed-point blockset. After verification of the algorithm, a fixed-point C program is developed for a specific fixed-point DSP processor. Software efficiency can be further improved by using mixed C-and-assembly programs, intrinsic functions, and optimized assembly routines in DSP libraries. This integrated software-development process enables students and engineers to understand and appreciate the important differences between floating-point simulations and fixed-point implementation considerations and applications.

Transition from Simulink to MATLAB in Real-Time Digital Signal Processing Education

W.S.Gan,Sen M Kuo
Journal Paper International Journal of Engineering Education, Vol. 21, No. 4, pp 587-595, 2005.

Abstract

In this paper, we propose a two-level approach for teaching digital signal processing (DSP) from basic concepts to the level of developing DSP software for real-time implementations on programmable DSP processors. In our approach, MATLAB and Simulink make the transition from theory to application easy and enjoyable. We use many interesting DSP demonstrations and examples for students to "see" the effects of signal processing in Simulink; and then ask students to "do" hands-on exercises in Simulink and MATLAB. The emphasis of "seeing" and "doing" can capture the students' attentions, cultivate their interests, and motivate their curiosities. This effective learning approach also allays fear of DSP that has been previously tagged as too theoretical and mathematically intensive.

FPGA Implementation of Parametric Loudspeaker System

K A Furi,W.S.Gan, Y K Chong
Journal Paper Microsystem and Microprocessor Journal, Vol. 28, pp 261-272, May 2004.

Abstract

Parametric loudspeaker system enables sound to be projected and directed to a specific listening area just like a beam of light. The advancement of Field Programmable Gate Array (FPGA) technology opens up a very interesting option for rapid implementation and easy configurable signal processing platform for parametric loudspeaker system. In this paper, the digital signal processing subsystem of the parametric loudspeaker system has been designed and implemented in FPGA platform using the Altera 1S10 device.

Powering next-generation multimedia apps with OMAP processor

F Kua, M T Wong,W.S.Gan
Journal Paper Electronic Engineering Times-Asia, Design Corner, pp32-38, 16-30 Jun 2003.

Abstract

S

Rapid Prototyping of DSP Algorithm in VLIW TMS320C6701 DSP

K H Hong, W.S.Gan
Journal Paper Journal of Microprocessors and Microsystems, Vol. 26, No. 7, pp 311-324, Sep 2002.

Abstract

In this paper, an overview of a rapid prototyping system using MATLAB Real-Time Workshop (RTW) and TI TMS320C6701-EVM is presented. The MATLAB RTW generated ANSI C code from Simulink blocksets has poor real-time implementation benchmarks for DSP kernels. To improve the performance of the DSP kernel blocksets on TI TMS320C6701 DSP, a design methodology employing various levels of optimising techniques are discussed and these algorithms are evaluated on the DSP. Modulo scheduling theory is applied to improve the loop performance. Performance results show that code developed based on optimal scheduling for the C6000 prototyping system is highly efficient. The optimised blockset provides an easy-to-use and highly efficient real-time DSP algorithm development platform.

Teaching and Learning the Hows and Whys of Real-Time Digital Signal Processing

W.S.Gan
Journal Paper IEEE Transaction on Education, Vol. 45, No. 4, pp336-343, Nov 2002.

Abstract

This paper discusses the approach taken in conducting a real-time digital signal processing (DSP) design class for undergraduates. The most recent TMS320C5402 DSP Starter Kit (DSK) from Texas Instruments (TI) is used, and a set of courseware that exposes students to both theoretical understanding and hands-on exercises is developed. It leads the students from the conceptual stage in DSP design to the actual implementation stage. This paper discusses the lecturer and students' perspective in this real-time DSP course.

An Integrated Environment for Rapid prototyping DSP Algorithms Using MATLAB and Texas Instruments’ TMS320C30

K H Hong, W.S.Gan
Journal Paper Microprocessors and Microsystems Journal, Vol. 24, No. 7, pp 349-3631, Nov. 2000.

Abstract

This paper presents the implementation of a rapid prototyping system, which involves the design of DSP algorithms using matlab Simulink blocksets, automated code generation, and downloading of executable code to the Texas Instruments’ Evaluation Module (TMS320C30-EVM). Various DSP algorithms were implemented and benchmarked in this system. It demonstrates that the matlab Simulink development system integrated with the TMS320C30-EVM provides a useful development tool for design verification of DSP algorithms. Performance results show that code developed using the rapid prototyping system is highly efficient and the development cycle time is greatly reduced, resulting in lower development cost.

Rapid Prototyping System for Teaching Real-Time Digital Signal Processing

W.S.Gan , Y K Chong, W Gong, W T Tan
Journal Paper IEEE Trans on Education, Vol. 43, No. 1, pp 19- 24, Feb 2000.

Abstract

A low-cost rapid-prototyping system using Texas Instruments' (TI) TMS320C30 Evaluation Module (EVM) based on the MathWorks development software is presented in this paper. The rapid prototyping system serves as an educational tool in learning digital signal processing (DSP) and seeing the concept realized in real time. The development software modules, Simulink, uses graphical block diagrams to create models for real-time implementation and the real-time workshop (RTW), generates C code to be downloaded onto the EVM. The entire building process is fully automatic. This includes compiling, assembling and downloading of the real-time algorithms. The system was found to be well suited for learning real-time DSP algorithms for both undergraduate and postgraduate levels.

A Low-Cost DSP-based Earmuff using Adaptive Active Noise Control

W.S.Gan, Y K Chong, M J Er
Journal Paper Microprocessor and Microsystems Journal, Vol 22, No.7, pp 412 – 422, Jan 1999.

Abstract

This paper presents the design and implementation of a digital signal processor (DSP)-based feed-forward narrow-band noise canceling system using the Analog Devices ADSP-2181 EZ-KIT Lite developer's kit. The theory of adaptive noise control (ANC) is applied to an ear-muff so that the entire system combines both passive and active techniques to reduce the harmful acoustic noise emitted by machinery and engines. The ANC developed uses a 100-tap finite impulse response (FIR) filter which encompasses the least mean square (LMS) and normalised LMS (NLMS) algorithms to adaptively `track' changing input conditions. The hardware and software aspects for real-time implementation, the signal conditioning circuits and the software developed for stereo multiplexing in the EZ-KIT Lite environment are presented. Performance results show significant improvements in noise attenuation levels when compared to use of a purely passive technique.

Editorial: Parametric Acoustic Array: theory, Advancement, and Applications

W.S.Gan , J Yang, T. Kamakura
Journal Paper Applied Acoustics (Special Issue on Parametric Acoustic Array), Vol. 73, No. 12, pp 1209-1210, Dec. 2012.

Abstract

In this review paper, we examine some of the recent advances in the parametric acoustic array (PAA) since it was first applied in air in 1983 by Yoneyama. These advances include numerical modelling for nonlinear acoustics, theoretical analysis and experimentation, signal processing techniques, implementation issues, applications of the parametric acoustic array, and some safety concerns in using the PAA in air. We also give a glimpse on some of the new work on the PAA and its new applications. This review paper gives a tutorial overview on some of the foundation work in the PAA, and serves as a prelude to the recent works that are reported by different research groups in this special issue.

A review of parametric acoustic array in air

W.S.Gan, J Yang, T Kamakura
Journal Paper Applied Acoustics, Vol. 73, No. 12, pp 1211- 1219, Dec. 2012.

Abstract

In this review paper, we examine some of the recent advances in the parametric acoustic array (PAA) since it was first applied in air in 1983 by Yoneyama. These advances include numerical modelling for nonlinear acoustics, theoretical analysis and experimentation, signal processing techniques, implementation issues, applications of the parametric acoustic array, and some safety concerns in using the PAA in air. We also give a glimpse on some of the new work on the PAA and its new applications. This review paper gives a tutorial overview on some of the foundation work in the PAA, and serves as a prelude to the recent works that are reported by different research groups in this special issue.

Identification of a parametric loudspeaker system using an adaptive Volterra filter

W Ji,W.S.Gan
Journal Paper Applied Acoustics, Vol. 73, No. 12, pp 1251-1262, Dec. 2012.

Abstract

Due to the parametric acoustic array effect in air, the input audible signal of a parametric loudspeaker system can be reproduced with high directivity at the target region. However, the reproduced audible signal suffers from harmonic distortion, which is the by-product of nonlinear interaction between the primary waves. In order to investigate this inherent nonlinear phenomenon, a nonlinear system identification model is developed based on an adaptive Volterra filter. Unlike the conventional loudspeaker, the nonlinear characteristic of a parametric loudspeaker system is dependent on several primary parameters in nonlinear acoustics, which include the initial pressure of the primary waves, the observing distance and angle, as well as ambient temperature and relative humidity. By using a truncated Volterra series up to the 2nd-order kernel, numerical simulations are conducted to develop a system model with one group of parameters and examine the quadratic nonlinear intensity for different parameters’ settings. Experimental measurements, which take into account of emitter’s response, are carried out to verify the modeling result and evaluate the model performance. Based on the Volterra system model, the sound pressure level and the harmonic distortion can be accurately predicted.

An alternative method to measure the on-axis difference-frequency sound in a parametric loudspeaker without using an acoustic filter

P.F Ji, W Liu, S. Wu, J. Yang, W.S.Gan
Journal Paper Applied Acoustics, Vol. 73, No. 12, pp 1244-1250, Dec. 2012

Abstract

The self-demodulation characteristic of finite-amplitude ultrasonic sound waves can be applied with parametric loudspeaker to reproduce audible sound with highly directivity. But measuring the difference-frequency sound is still a problem due to the spurious sound generated as a result of nonlinearity caused by the product of the primary waves at the receiving system. In this paper, based on the phase-cancellation method and the Gaussian beam expansion technique, an alternative method is proposed to measure the on-axis difference-frequency sound accurately without using any traditional acoustic filter, where the spurious sound can be greatly reduced or even eliminated. The proposed method is more suitable for the case where the piston source in the parametric loudspeaker comprises multiple small piezoelectric transducers (PZTs) and each transducer element in the array may have different frequency response. The validity of the proposed method is confirmed both by simulations and experiments.

Analysis and calibration of system errors in steerable parametric loudspeakers

C. Shi,W.S.Gan
Journal Paper Applied Acoustics, Vol. 73, No. 12, pp 1263-1270, Dec. 2012.

Abstract

By adjusting a set of delay amounts and amplitudes of the ultrasonic transducer (primary source) array in parametric loudspeakers, the directional sound beam can be steered within a range of predefined angles. This beamsteering characteristic of parametric loudspeakers has been proposed in theory and validated by measurements. In particular, the locations of the mainlobe and grating lobes can be predicted within a certain degree of accuracy in theory. However, errors incur in different stages of implementation. Thus, mismatches are observed between theoretical and measured beampatterns. In this paper, four types of system errors are analyzed for the primary-frequency waves and the difference-frequency waves based on the phased array theory and the product directivity principle, respectively. The degraded beampatterns which are caused by system errors are analyzed and calibrated by using a combined optimization approach of Monte Carlo method and nonlinear least squares method. Experimental results are presented to show the advantage of the proposed calibration method that leads to significant reduction of mismatch between theoretical and measured beampatterns at both the primary frequency and the difference frequency.

Product directivity models for parametric loudspeakers

C. Shi,W.S.Gan
Journal Paper J. Acoust. Soc. Am., Vol. 131, No. 3, pp 1938-1945, March 2012.

Abstract

In a recent work, the beamsteering characteristics of parametric loudspeakers were validated in an experiment. It was shown that based on the product directivity model, the locations and amplitudes of the mainlobe and grating lobes could be predicted within acceptable errors. However, the measured amplitudes of sidelobes have not been able to match the theoretical results accurately. In this paper, the original theories behind the product directivity model are revisited, and three modified product directivity models are proposed: (i) the advanced product directivity model, (ii) the exponential product directivity model, and (iii) the combined product directivity model. The proposed product directivity models take the radii of equivalent Gaussian sources into account and obtain better predictions of sidelobes for the difference frequency waves. From the comparison between measurement results and numerical solutions, all the proposed models outperform the original product directivity model in terms of selected sidelobe predictions by about 10 dB..

Audio Projection: Directional sound and its application in immersive communication

W.S.Gan , E. L. Tan, Sen M Kuo
Journal Paper IEEE Signal Processing Magazine, Vol. 28, No. 1, pp 43-57, Jan 2011. (Special Issue on Immersive Communication)

Abstract

The parametric loudspeaker provides an effective means of projecting sound in a highly directional manner without using large loudspeaker arrays to form sharp directional beams. It can be augmented with conventional loudspeakers to create a more immersive audio soundscape. Deployment of parametric loudspeakers in many public places where private messaging can make a difference in attracting attention, conveying messages without needing headphones, and creating private listening zones to reduce noise pollution. Digital signal processing plays a significant role in enhancing the aural quality of the parametric loudspeakers, and array processing can help to shape and steer the beam electronically. In addition, other signal processing techniques can also be applied to add more flexibility and improve the performance of parametric loudspeakers. These developments rely heavily on the latest techniques in acoustics and audio signal processing to overcome some of the current limitations in nonlinear acoustics modeling and ultrasonic transducers' technology. A useful feature in sound projection is to realize a highaccuracy digital beamsteering capability in air using an array of parametric loudspeakers. An in-depth study into the theoretical model of wave steering capability in parametric array in air can provide some hints on how we can best steer the demodulated signal in an efficient manner. As seen from this article, digital signal processing provides the main engine to achieve directional sound projection, and new digital processing techniques will be devised to provide a better quality, controllable audio beaming, and efficient sound focusing device in the future.

Grating Lobe Elimination In steerable Parametric Loudspeaker

C Shi,W.S.Gan
Journal Paper IEEE Transaction on Ultrasonics, Ferroelectrics, and Frequency Control, Vol. 58, No. 2, pp 437-450, Feb 2011.

Abstract

In the past two decades, the majority of research on the parametric loudspeaker has concentrated on the nonlinear modeling of acoustic propagation and pre-processing techniques to reduce nonlinear distortion in sound reproduction. There are, however, very few studies on directivity control of the parametric loudspeaker. In this paper, we propose an equivalent circular Gaussian source array that approximates the directivity characteristics of the linear ultrasonic transducer array. By using this approximation, the directivity of the sound beam from the parametric loudspeaker can be predicted by the product directivity principle. New theoretical results, which are verified through measurements, are presented to show the effectiveness of the delay-and-sum beamsteering structure for the parametric loudspeaker. Unlike the conventional loudspeaker array, where the spacing between array elements must be less than half the wavelength to avoid spatial aliasing, the parametric loudspeaker can take advantage of grating lobe elimination to extend the spacing of ultrasonic transducer array to more than 1.5 wavelengths in a typical application.

“A comparative analysis of preprocessing methods for the parametric loudspeaker based on the Khokhlov-Zabolotskaya-Kuznetsov equation for speech reproduction

P.F. Ji, E. L Tan, W.S.Gan, Jun Yang
Journal Paper IEEE Transaction on Audio, Speech, and Language Processing, Vol. 19, No. 4, pp 937-946, 4 May 2011.

Abstract

Based on the Berktay's farfield solution, various preprocessing methods were proposed to reduce the distortion of the highly directional audible signal in the parametric loudspeaker. However, the Berktay's farfield solution is an approximated model of nonlinear acoustic propagation. To determine the effectiveness of these methods, we analyze various preprocessing methods theoretically for directional speech reproduction using the Khokhlov-Zabolotskaya-Kuznetsov (KZK) equation, which provides a more accurate model of nonlinear acoustic propagation. In order to reduce the distortion effectively in the parametric loudspeaker with these preprocessing methods, the initial sound pressure level of the carrier frequency is found to be less than 132 dB according to the KZK equation. Unlike the Berktay' farfield solution that results in a +12 dB/octave gain slope, different gain slopes are derived using the KZK equation and appropriate equalizers are proposed to improve the frequency response of the parametric loudspeaker. The optimal preprocessing method for directional speech reproduction is established based on the KZK equation, which has a relatively flat frequency response of the desired speech signal and the best total harmonic distortion performance.

On Preprocessing Techniques for Bandlimited Parametric Loudspeakers

P.F. Ji, E. L Tan, W.S.Gan
Journal Paper Applied Acoustics, Vol. 71, No. 5, pp 486-492, May 2010.

Abstract

The self-demodulation property of finite-amplitude ultrasonic waves can be applied with parametric loudspeaker to produce audible sound. A special characteristic of the reproduced sound waves using parametric loudspeaker is its high directivity. However, the demodulated signal from parametric loudspeaker suffers from high distortion. To reduce the distortion in the demodulated signal, preprocessing of the modulating signal is usually employed. To determine the effectiveness of the preprocessing technique, an important practical constraint on the bandwidth of the ultrasonic transducer of the parametric loudspeaker should be accounted. In this paper, we shall discuss a class of preprocessing techniques that is based on quadrature amplitude modulation. As compared to the conventional preprocessing methods used with bandlimited ultrasonic transducer, the demodulated signal from our proposed preprocessing techniques exhibits lower distortion.

Development of Parametric Loudspeaker: A Novel Directional Sound Generation Technology

C Shi, W.S.Gan
Journal Paper IEEE Potentials, Vol. 29, No. 6, pp 20-24, Nov/Dec 2010.

Abstract

Development of Parametric Loudspeaker: A Novel Directional Sound Generation Technology

The Investigation of Localized Sound Generation Using Two Ultrasound Beams

P.F. Ji, J. Yang, W.S.Gan
Journal Paper IEEE Transaction on Ultrasonics, Ferroelectrics, and Frequency Control, Vol. 56, No. 6, pp 1282-1287, Jun/Jul 2009

Abstract

A quantitative analysis of the effects of difference frequency, source separation, and crossing angle on the generated scattered difference frequency sound fields is presented to evaluate the feasibility of localized sound production using 2 uniform pistons. Nonlinear crossed beam experiments were also carried out in an anechoic chamber. Experimental results show that the audible sound could be generated within the interaction region defined by the overlap volume of 2 ultrasonic beams.

Bandwidth-efficient Recursive pth-order Equalization for Correcting Baseband Distortion in Parametric Loudspeakers/h4>
Kelvin Lee,W.S.Gan, Jun Yang
Journal Paper IEEE Transaction on Audio, Speech and Language Processing, Vol. 14, No. 2, pp 706 – 710, Mar 2006.

Abstract

A bandwidth-efficient recursive implementation of pth-order equalization is developed in order to correct the inherent baseband distortion in parametric loudspeakers. Assuming that the nonlinear system is largely quadratic, a substitution of it can be made by using an analytic pair of all-pass sections cascaded with a squared magnitude block. The all-pass sections comprise of a delay block and an approximate linear-phase infinite impulse response (IIR) Hilbert transformer to give a Hilbert transform pair. In this way, the distortion terms produced are due to the difference frequencies only and therefore within the bandwidth of the original input signal. When used together with a single-sideband (SSB) amplitude modulation (SSB-AM) scheme, this new method allows the equalized SSB output signal to be faithfully reproduced to correct the distortion. Simulation results in this paper show that the new method is able to suppress residual in-band distortion components by -70 dB or lower.

A Digital Beamsteerer for Difference Frequency in Parametric Array

W.S.Gan, J Yang, K S Tan, M H Er
Journal Paper IEEE Transaction on Audio, Speech and Language Processing, Vol. 14, No. 3, pp 1018 – 1025, May 2006.

Abstract

A steerable audio system can be realized using parametric array. However, the available steerable angle is often limited by the sampling interval used in the digital system. As such, the smallest steerable angle is large (∼26°) for several hundred kilohertz of sampling frequency. Although there are some fractional delay or frequency domain algorithms can be used to improve the steering angle, most of the algorithms are either computational intensive or introduce error during the process. In this paper, an algorithm is proposed to rectify this problem by applying separate delays to the carrier and sideband frequencies. Different weighting functions also added to the carrier and sideband frequencies to control the difference frequency's beamwidth and sidelobe. Most importantly, the proposed system can steer the difference frequency to a small angle with minimal computation.

Beamwidth control in a parametric acoustic array

J. Yang, W.S.Gan , K.S. Tan, and M. H. Er
Journal Paper Japan Journal of Applied Physics, Vol. 44, No. 9A, pp 6817-6819, 2005.

Abstract

Directional sound can be generated by amplitude-modulating the ultrasound carrier wave with an audio signal and then transmitting it from an array system. On the basis of the analysis of its radiation pattern, an algorithm using delay-and-sum beamforming is proposed to control the beamwidth and sidelobe of demodulated audible sound. Simulation results show that enhanced performance in terms of sidelobe level can be achieved by using the Chebyshev weighting function, compared with the beamforming algorithm without shading.

Acoustic beamforming of a parametric speaker comprising ultrasonic transducers

J. Yang, W.S.Gan , K.S. Tan, and M. H. Er
Journal Paper Sensors and Actuators A: Physical, Vol. 125, No. 1 pp 91-99, 2005.

Abstract

A directional audible sound can be generated in air by means of the nonlinear interaction between intense amplitude modulated (AM) ultrasonic waves, which attracts much attention in the audio industry. In the case of sound reproduction by a parametric speaker comprising an array of ultrasonic transducers, a novel algorithm with Chebyshev window has been proposed to control the sidelobe level of the beam pattern by utilizing the acoustic nonlinearity and array signal processing technique. Furthermore, the beamforming approach is extended to the design of a broadband beamformer. In this paper, a single sideband modulation (SSB) system is developed for transmitting audio signal due to its efficient use of both power and bandwidth. By adding different weightings to the carrier and sideband frequencies, a constant beamwidth beamforming is achieved and confirmed in the simulation experiments.

Modeling of Finite-amplitude Sound Beams: Second Order Fields Generated by a Parametric loudspeaker

J Yang, K Sha,W.S.Gan , J Tian
Journal Paper IEEE Transaction on Ultrasonics, Ferroelectrics, and Frequency Control, Vol. 52, No. 4, pp 610-618, Apr 2005.

Abstract

The nonlinear interaction of sound waves in air has been applied to sound reproduction for audio applications. A directional audible sound can be generated by amplitude-modulating the ultrasound carrier with an audio signal, then transmitting it from a parametric loudspeaker. This brings the need of a computationally efficient model to describe the propagation of finite-amplitude sound beams for the system design and optimization. A quasilinear analytical solution capable of fast numerical evaluation is presented for the second-order fields of the sum-, difference-frequency and second harmonic components. It is based on a virtual-complex-source approach, wherein the source field is treated as an aggregation of a set of complex virtual sources located in complex distance, then the corresponding fundamental sound field is reduced to the computation of sums of simple functions by exploiting the integrability of Gaussian functions. By this result, the five-dimensional integral expressions for the second-order sound fields are simplified to one-dimensional integrals. Furthermore, a substantial analytical reduction to sums of single integrals also is derived for an arbitrary source distribution when the basis functions are expressible as a sum of products of trigonometric functions. The validity of the proposed method is confirmed by a comparison of numerical results with experimental data previously published for the rectangular ultrasonic transducer.

Nonlinear wave propagation for a parametric loudspeaker

J Yang, K Sha,W.S.Gan , J Tian
Journal Paper IEICE Trans. Fundamentals, Vol. E87-A, No.9, pp 2395-2400, 2004.

Abstract

A directional audible sound can be generated by amplitude-modulated (AM) into ultrasound wave from a parametric array. To synthesize audio signals produced by the self-demodulation effect of the AM sound wave, a quasi-linear analytical solution, which describes the nonlinear wave propagation, is developed for fast numerical evaluation. The radiated sound field is expressed as the superposition of Gaussian Beams. Numerical results are presented for a rectangular parametric loudspeaker, which are in good agreement with the experimental data published previously.

A simple calculation method for the self and mutual radiation impedance of flexible rectangular patches in a rigid infinite baffle

K Sha, J Yang, W.S.Gan
Journal Paper Journal of Sound and Vibration, Vol. 282, No. 1-2, pp 179-195, 6 April 2005.

Abstract

A numerical model has been developed to calculate the self- and mutual-radiation impedance in the cases of uniformly and flexibly vibrating rectangular patches in a rigid infinite baffle. The spatial convolution approach is employed here to derive general expressions for the radiation impedance of a rectangular radiator in the form of simple integrals, which allows a fast evaluation numerically. The presented integral solution agrees with that obtained for the mutual-radiation impedance of a uniformly vibrating rectangular piston by the use of the classical approach. The numerical results of self-radiation impedance of a square piston are compared with the tabulated values published previously. As examples of flexibly vibrating rectangular patch, a closed-form expression is first given for the radiation impedance in the normal mode of vibration. The numerical results reveal that the computation time in obtaining accurate calculations is greatly reduced by using the proposed method.

Radiation Impedance Calculation for Arbitrary Shaped Piston

J Yang, K Sha,W.S.Gan , J Tian
Journal Paper Japan Journal of Applied Physics, Vol. 43, No. 9A, pp 6274-6277, 2004.

A general solution for determining self- and mutual-radiation impedances for rectangular radiators is presented in a closed form with a double integral. Using this solution, an efficient calculation method is developed for the radiation impedance of an arbitrarily shaped piston in a rigid infinite baffle. The proposed algorithm is verified by its agreement with previously published results for square pistons. Its utility is demonstrated with an example of radiation impedance of a circular piston..

A complex virtual source approach for calculator the diffraction beam field generated by a rectangular planar source

K Sha, J Yang,W.S.Gan , J Tian
Journal Paper IEEE Transaction on Ultrasonics, Ferroelectrics, and Frequency Control, Vol. 50, No. 7, pp 890-897, Jul 2003

Abstract

In this paper, a complex virtual source approach for calculating the ultrasound field generated by a rectangular planar source is presented. Instead of using a real rectangular plane source, the equivalent sources that have complex amplitudes in complex space are used to compute the sound field distribution. The parabolic equation first is solved in the k-space domain by applying Fourier transform. The k-space domain source is then expressed as a set of Gaussian functions, and the related coefficients is determined by the optimization method. The analytic solution then is derived, and the effect of the parameters on the calculation accuracy is discussed. The comparison between the proposed fast numerical scheme and previous methods (Fresnel integral and Ocheltree's method) and are given in an example. The numerical results reveal that the computation time in obtaining accurate calculations is greatly reduced by using the proposed method.

Time-domain lifted wavelet collocation method for modeling nonlinear wave propagation

C M Lee,W.S.Gan , J Tian
Journal Paper Acoustics Research Letter On-line (ARLO), Vol.3, No.4, pp124-129, Oct 2002.Chinese Physics Letters, Vol. 21, No. 1, pp 110-113, 2004.

Abstract

A time-domain adaptive numerical method for modeling nonlinear wave propagation is developed. This method is based on a second-generation wavelet collocation using a lifting scheme and makes use of the multilevel decomposition nature of the scheme to allow for automatic grid refinement according to the magnitude of waveform steepening. The multiplication in the nonlinear term is also easy due to the collocation nature. With thresholding, the solution is compact at every level of resolution and computed only at collocation points associated with the remaining significant wavelet coefficients. The error tolerance and compression ratio of the new method are totally controlled by the threshold value used. This brings substantial savings in computation time when compared to the conventional finite difference scheme on a uniformly fine grid.

Linear Estimation Based Primary-Ambient Extraction for Stereo Audio Signals

J. J. He, E.L. Tan,W.S.Gan
Journal Paper IEEE/ACM Transaction on Audio, Speech, and Language Processing, Vol. 22, Issue No. 2, pp 505 -517, Feb 2014.

Abstract

Audio signals for moving pictures and video games are often linear combinations of primary and ambient components. In spatial audio analysis-synthesis, these mixed signals are usually decomposed into primary and ambient components to facilitate flexible spatial rendering and enhancement. Existing approaches such as principal component analysis (PCA) and least squares (LS) are widely used to perform this decomposition from stereo signals. However, the performance of these approaches in primary-ambient extraction (PAE) has not been well studied and no comparative analysis among the existing approaches has been carried out so far. In this paper, we generalize the existing approaches into a linear estimation framework. Under this framework, we propose a series of performance measures to identify the components that contribute to the extraction error. Based on the generalized linear estimation framework and our proposed performance measures, a comparative study and experimental testing of the linear estimation based PAE approaches including existing PCA, LS, and three proposed variant LS approaches are presented.

Perceptually-Motivated Objective Grading of Nonlinear Processing in Virtual Bass Systems

N Oo,W.S.Gan
Journal Paper Journal of Audio Engineering Society, Vol. 59, No. 11, pp 804-824, Nov 2011.

Abstract

When size limitations of loudspeakers prevent the reproduction of low-frequency sounds, nonlinear devices can be used to create the illusion of the missing bass. Such virtual bass systems generate harmonics of the missing fundamental. However, they also can generate unwanted intermodulation distortion, which appears to be dependent on the particular audio sample and the selected nonlinearity. A detailed analysis showed that the ideal nonlinearity should not be even-symmetric, and its second derivative should be less than zero on the input interval 0 to 1.

Time-Reversal Approach to the Stereophonic Acoustic Echo Cancellation Problem,”

D. Q. Nguyen,W.S.Gan , Andy W.H. Khong
Journal Paper IEEE Transaction on Audio, Speech, and Language Processing, Vol. 19, No. 2, pp 385-395, Feb 2011.

Abstract

Stereophonic acoustic echo cancellation (SAEC) plays an important role in delivering realistic teleconferencing experience. The fundamental problem of SAEC system is that stereophonic channels are linearly related and this results in slow convergence of the adaptive filters. In this paper, we present a novel algorithm by employing a selective time-reversal block to solve the SAEC problem which results in a significant increase in the convergence performance of adaptive filters such that the stereophonic image as well as quality are preserved. The proposed algorithm employs time-reversal operation on selective blocks of input data samples for one of the two channels to decorrelate stereophonic channels in the SAEC system. To achieve good stereophonic perception, time-reversal operation is only applied to the selective blocks whose magnitudes fall below a pre-determined threshold. Theoretical and numerical simulation results are also studied and investigated to show that the proposed algorithm achieves faster convergence in terms of normalized misalignment and better stereophonic perception with less audio distortion compared to the well-known nonlinear transformation algorithm for the SAEC system.

The DORT Solution in Acoustic Inverse Scattering Problem of a Small Elastic Scatterer

D. Q. Nguyen,W.S.Gan
Journal Paper Elsevier Journal of Ultrasonics, Vol. 50, pp 829-840, 2010.

Abstract

The DORT (French acronym for Décomposition de l'Opérateur de Retournement Temporel) method is a novel approach for active detection and focusing of acoustic waves on the targets in the scattering medium. This technique involves the determination of the invariant of the time-reversal operator obtained by measurement of the scattering data in a pulse-echo mode. In this paper, a proposed approach based on the DORT method is developed to solve the acoustic inverse scattering problem of a small metallic scatterer. The proposed approach not only estimates the position of the scatterer, but also determines the physical properties of an unknown metallic scatterer such as the shape (cylinder or sphere), the material (density), and the size (radius) in an anisotropic scattering case. Theoretical and numerical simulation results are also studied and investigated to show that the proposed approach can simultaneously characterize all those properties of an unknown metallic scatterer. Moreover, the advantage of the proposed approach is to avoid the complex iterative scheme in solving the direct scattering problem and results in smaller computational load and faster implementation.

Editorial for Intelligent Audio, Speech, and Music Processing Applications

W.S.Gan S. M. Kuo, J. H. L. Hansen,
Journal Paper EURASIP Journal on Audio, Speech, and Music Processing, Vol. 2008, Article ID 854716, 2 pages, 2008.

Abstract

Future audio, speech, and music processing applications need innovative intelligent algorithms that allow interactive human/environmental interfaces with surrounding devices/systems in real-world settings to control, process, render, and playback/project sound signals for different platforms under a diverse range of listening environments. These intelligent audio, speech, and music processing applications create an environment that is sensitive, adaptive, and responsive to the presence of users. Three areas of research are considered in this special issue: analysis, communication, and interaction. Analysis covers both preprocessing of sound signals and extraction of information from the environment. Communication covers the transmission path/network, coding techniques, and conversion between spatial audio formats. The final area involves intelligent interaction with the audio/speech/music environment based on the users' location, signal information, and acoustical environment.

Novel DORT Method in Non-Well-Resolved Scatterer Case

D. Q. Nguyen,W.S.Gan
Journal Paper IEEE Signal Processing Letters, Vol.15, pp705 – 708, 2008.

Abstract

The decomposition of time reversal operator (DORT) method is an efficient technique to select focusing signal on the target in well-resolved scatterer case. The DORT method requires the measurement of the inter-element impulse responses and diagonalizes these responses to find eigenstructure of the medium. However, the DORT method is not able to produce satisfactory result for non-well-resolved scatterer case (i.e., two scatterers that are placed closely about 0.2lambda). Therefore, in this letter, a weighted least squares (WLS) algorithm is introduced in the DORT method to perform selective focusing in non-well-resolved scatterer case. This technique called the WLS-DORT method performs better spatial focusing resolution than the conventional DORT method and results in simpler computational complexity than the minimum variance beamforming method under practical implementation.

Initial value independent optimization for flat-top power pattern synthesis using non-uniform linear arrays

Y. Wen, W.S.Gan, J. Yang
Journal Paper IEE Electronics Letters, Vol. 41, No. 12, pp 677 – 678, 9 June 2005.

Abstract

A multi-stage optimisation method for solving the nonlinear least square error problem in flat-top power pattern synthesis using non-uniform linear arrays is presented. The optimisations are conducted from low to high order in eigenspace, and without the requirement of multidimensional initial value. Simulations show the robustness of the method over a conventional method dependent on initialisation.

Nonlinear Least-square Solution to Flat-top Pattern Synthesis Using Arbitrary Linear Array

Y. Wen, W.S.Gan, J. Yang
Journal Paper Signal Processing, Vol. 85, pp 1869-1874, 2005.

Abstract

This paper presents a new approach for synthesizing flat-top patterns, based on the least-square error criterion. The cost function is formulated according to the amplitude approximation error without phase constraint. The optimal array weight is obtained by using the Levenberg–Marquardt nonlinear optimization algorithm. Simulations are performed to compare the proposed approach with the Woodward–Lawson method and two recent methods using minimax and adaptive array techniques, respectively. The results indicate that the approach is effective in sidelobe control and synthesizing prespecified patterns for arbitrary linear arrays.

Strategies for an acoustical-hotspot generation

Y. Wen, J. Yang, W.S.Gan
Journal Paper IEICE Trans. Fundamentals, Vol.E88-A, No.7, Jul 2005.

Abstract

Two methods for hotspot generation using multiple sources, known as time-delay (TD) method and maximum-control-gain (MCG) method are investigated in the two typical acoustical fields, namely, the free field and a rectangular room. Based on the theoretical analysis and simulations, strategies are developed according to the sound field where the target region is defined. In the free field, the MCG method can be used if the performance in terms of control gain is the priority for an optimal control, whereas the TD method is more preferable if the simplicity of im- plementation is the first consideration. In a room environment, if a target region is defined in the near field where the direct sound dominates, the TD method is still effective. However, in the far field where the reverberant sound prevails, only the MCG method is applicable. The near field/far field can be roughly separated according to the critical distance from the sources in the room.

A fast algorithm for the sound projection using multiple sources

Y. Wen, W.S.Gan, J. Yang
Journal Paper IEICE Trans. Fundamentals, Vol. E88-A, No. 7, Jul 2005.

Abstract

An algorithm for the sound projection using multiple sources is presented. The source strength vector is obtained by using a fast estimation approach instead of the conventional eigenvalue decomposition (EVD) method. The computation load is therefore greatly reduced, which makes the algorithm more efficient in practical applications.

Development of Virtual Sound Imaging System using Triple Elevated Speakers

J Yang,W.S.Gan, S E Tan
Journal Paper IEEE Transaction on Consumer Electronics, Vol. 50, No. 3, pp 916-922, Aug 2004.

Abstract

An enhanced virtual sound imaging system using triple elevated-speaker projection is proposed. Based on the robustness analysis, an efficient auditory image correction scheme is developed to improve system performance. This is achieved by specially designed inverse filter together with HRTF filtering. The sound imaging algorithm is implemented on a TMS320C6201 EVMDSP board and subjective testing is also carried out to demonstrate the virtual sound effects.

Robust Regularization for Enhanced Virtual Sound Imaging

J Yang,W.S.Gan
Journal Paper IEICE Transaction on Fundamentals of Electronics, Communications, Computer Sciences, Vol. E86-A, No. 8, pp 2061-2062, Aug 2003.

Abstract

This letter outlines a scheme to produce a wider robust bandwidth, with better approximations to the perfect re- production of pre-recorded acoustic signals. Multi-parameter in- verse filtering method is proposed in the virtual sound imaging system for improving the robustness performance. The superior- ityof this new type of inverse filter is demonstrated on a 3-speaker system.

Improved Sound Separation using three loudspeakers

J Yang, S E Tan,W.S.Gan
Journal Paper Acoustics Research Letter On-line (ARLO), Vol. 4, No. 2, pp 47-52, Apr 2003.

Abstract

In a virtual soundimaging system, crosstalk cancellation filters are used to create an effective sweet spot for 3D sound reproduction via multiple loudspeakers. A new 3-channel system is proposed to improve system performance on sound separation. Based on the robustness analysis of a crosstalk canceller, a modified-inverse filter technique is explored and demonstrated using two different examples of symmetric speaker positions. The simulation results indicate that the present system is robust over a wider bandwidth compared to a conventional 2-channel system.

Virtual Bass for Home Entertainment, Multimedia PC, Game Station and Portable Audio Systems

W.S.Gan , S.M. Kuo, C.W. Toh
Journal Paper IEEE Trans. on Consumer Electronics, Vol. 47, No. 4, pp 787-794, Nov 2001.

Abstract

In digital audio playback systems for home entertainment, multimedia PC, game station and portable audio devices, there is a strong demand to produce deep bass using small multimedia speakers and earphones, without the need for additional subwoofer or expensive speakers/earphones. This paper describes a technique in synthesizing psycho-acoustic bass over a pair of stereo speakers using the pitch perception theory. This new method of creating a bass effect for the musical signal has been found to be very effective and simple in implementation, without increasing significant cost in many modern digital audio systems

Elevated Speaker Projection for Digital Home Entertainment System

W.S.Gan, S E Tan, M H Er, Y K Chong
Journal Paper IEEE Trans. on Consumer Electronics, Vol. 47, No. 3, pp 631 – 637, Aug 2001.

Abstract

This paper presents a novel processing technique to enhance the sound images when playback over elevated loudspeakers for both two-speaker and multiple-speaker configuration. This is achieved by specially designed inverse filter together with HRTF filtering

Application of virtual bass in audio crosstalk cancellation.

S E Tan, W.S.Gan
Journal Paper IEE Electronics Letters, Vol.36, No.17, pp1500-1501,17 Aug 2000

Abstract

At low frequency, the interaural level difference (ILD) for near sources is significantly large. Crosstalk cancellation for low-frequency signals is difficult and generally will not result in the required ILD for reproduction of near source signals. A `virtual bass' technique is proposed that creates a perceptually equivalent bass of the signal. This technique translates the low-frequency signal to a higher frequency band, thereby avoiding crosstalk cancellation at low frequency

Direct Concha Excitation for Introduction of Individualized Hearing Cues

C J Tan,W.S.Gan
Journal Paper Journal of Audio Engineering Society, Vol. 48, No. 7/8, pp 642-653, Jul/Aug 2000.

Abstract

Nonindividualized head-related transfer functions (HRTFs) are known to cause significant front-back and elevation confusions when used for three-dimensional sound systems. In addition headphones tend to induce in-head localization and rear perceptual bias. A method of concha excitation is proposed as a means of introducing individual cues without the need of individual HRTF measurements, which may overcome some of these difficulties. A prototype headphone was designed and tested.

Speaker placement for robust virtual audio display system

J Yang, W.S.Gan
Journal Paper IEE Electronic Letters, Vol. 36 No.7, pp683-685, 30 Mar 2000.

Abstract

Based on a robustness analysis of the multichannel crosstalk canceller, a physical insight into a 3D audio system is presented, and hence a novel method for placing multiple speakers is introduced to increase the robustness of a sound reproduction system. Furthermore, the optimum loudspeaker positions are derived and simulation results are presented

Wavelet packet decomposition for spatial sound conditioning

C J Tan, W.S.Gan
Journal Paper IEE Electronics Letters, Vol.35, No.21, pp1821-1823, 14 Oct 1999.

Abstract

It is well known that sound localisation is heavily dependent on frequency. Band-limited signals generally suffer from poor localisation potential. While significant effort has been invested in HRTF modelling, room acoustics and crosstalk filtering, research into alleviating the spatial system performance dependence on source spectral characteristics has been largely overlooked. The authors briefly consider how one can compensate source frequency-deficient bands that are critical to localisation with wavelet packet decomposition

User-defined spectral manipulation of HRTF for improved localization in 3D sound systems

C J Tan,W.S.Gan
Journal Paper IEE Electronics Letters, Vol.34, No.25, pp2387-2389, Dec 1998.

Abstract

Front-back and elevation confusions typically occur when non-individualised HRTFs (head related transfer functions) are used in a 3D sound system. A practical strategy is proposed to allow more `customisation' of the 3D sound system. Thus it envisioned that the performance of the basic 3D sound system would be enhanced with the flexibility of user-defined adjustments without the actual measurement of an individual's HRTF

Subband Adaptive Filtering: Theory and Implementation

K.A. Lee, W.S.Gan, S.M. Kuo
Book West Sussex: Wiley, 2009 | ISBN: 0470516941 0
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Description

Subband adaptive filtering is rapidly becoming one of the most effective techniques for reducing computational complexity and improving the convergence rate of algorithms in adaptive signal processing applications. This book provides an introductory, yet extensive guide on the theory of various subband adaptive filtering techniques. For beginners, the authors discuss the basic principles that underlie the design and implementation of subband adaptive filters. For advanced readers, a comprehensive coverage of recent developments, such as multiband tap–weight adaptation, delayless architectures, and filter–bank design methods for reducing band–edge effects are included. Several analysis techniques and complexity evaluation are also introduced in this book to provide better understanding of subband adaptive filtering. This book bridges the gaps between the mixed–domain natures of subband adaptive filtering techniques and provides enough depth to the material augmented by many MATLAB® functions and examples.

Key Features:

Acts as a timely introduction for researchers, graduate students and engineers who want to design and deploy subband adaptive filters in their research and applications. Bridges the gaps between two distinct domains: adaptive filter theory and multirate signal processing. Uses a practical approach through MATLAB®-based source programs on the accompanying CD. Includes more than 100 M-files, allowing readers to modify the code for different algorithms and applications and to gain more insight into the theory and concepts of subband adaptive filters. Subband Adaptive Filtering is aimed primarily at practicing engineers, as well as senior undergraduate and graduate students. It will also be of interest to researchers, technical managers, and computer scientists.

Contents

  • About the authors.
  • Preface.
  • Acknowledgments.
  • List of symbols.
  • List of abbreviations.
  • 1. Introduction to adaptive filters.
  • 2. Subband decomposition and multirate systems.
  • 3. Second-order characterization of multirate filter banks.
  • 4. Subband adaptive filters.
  • 5. Critically sampled and oversampled subband structures.
  • 6. Multiband-structured subband adaptive filters.
  • 7. Stability and performance analysis.
  • 8. New research directions.
  • Appendix A Programming in MATLAB.
  • Appendix B Using MATLAB for adaptive filtering and subband adaptive filtering.
  • Appendix C Summary of MATLAB scripts, functions, examples and demos.
  • Appendix D Complexity analysis of adaptive algorithms.
  • Index

Active noise control using a functional link artificial neural network with the simultaneous perturbation learning rule

Y.L. Zhou, Q.Z. Zhang, T.Zhang, X.D. Li,W.S.Gan
Journal Paper Journal of Shocks and Vibration, Vol. 16, No. 3, pp 325-334, 2009.

Abstract

In practical active noise control (ANC) systems, the primary path and the secondary path may be nonlinear and time-varying. It has been reported that the linear techniques used to control such ANC systems exhibit degradation in performance. In addition, the actuators of an ANC system very often have nonminimum-phase response. A linear controller under such situations yields poor performance. A novel functional link artificial neural network (FLANN)-based simultaneous perturbation stochastic approximation (SPSA) algorithm, which functions as a nonlinear mode-free (MF) controller, is proposed in this paper. Computer simulations have been carried out to demonstrate that the proposed algorithm outperforms the standard filtered-x least mean square (FXLMS) algorithm, and performs better than the recently proposed filtered-s least mean square (FSLMS) algorithm when the secondary path is time-varying. This observation implies that the SPSA-based MF controller can eliminate the need of the modeling of the secondary path for the ANC system.

Convergence Analysis of Narrowband Active Noise Equalizer System under Imperfect Secondary Path Estimation

L. Wang, W.S.Gan
Journal Paper IEEE Transaction on Audio, Speech and Language Processing, Vol. 17, No. 4, pp 566-571, May 2009.

Abstract

Active noise equalizer systems are used to adjust the noise level in an environment, based on the preference of retaining noise information. Several researches have been carried out to determine the maximum step size bond of narrowband active noise control system with perfect secondary path estimation without gain factor consideration. However, in practical environment, secondary path estimation error of the system exists. In this paper, a stochastic approach analysis is applied to determine the maximum step size of the system under imperfect secondary path estimation. Simulation results are conducted to verify the analysis. Results show that the gain factor, sampling frequency, and secondary path estimation errors are all major factors governing the maximum step size of the narrowband active noise equalizer system under imperfect secondary path estimation.

On the use of an SPSA-based model-free feedback controller in active noise control for periodic disturbances in a duct

Y.L. Zhou, Q.Z. Zhang, X.D. Li, W.S.Gan
Journal Paper Journal of Sound and Vibration, Vol. 317, No. 3-5, pp 456-472, 11 Nov 2008.

Abstract

In this paper, a feedback active noise control (FANC) system using a model-free (MF) controller based on simultaneous perturbation stochastic approximation (SPSA) algorithm is considered. The structure of the FANC system is first described, and the SPSA algorithm is briefly reviewed. Subsequently, the SPSA-based MF control algorithm employed in the FANC system is derived. Computer simulations are carried out to suggest that the SPSA-based MF control algorithm is effective for ANC system and much more efficient than the finite-difference stochastic approximation (FDSA) algorithm. The controller based on the SPSA-based MF control algorithm is implemented on the Texas Instruments digital signal processor (DSP) TMS320VC33. Experimental results show that the proposed scheme is able to significantly reduce periodic disturbances without the need to model the secondary path. At the same time, the simulations and the experimental verification tests also show that the convergence rate of the SPSA-based MF control algorithm is acceptable, and the SPSA-based MF control algorithm has better tracking ability under variable secondary path. This observation implies that the SPSA-based MF controller eliminates the need of modeling of the secondary path for the FANC system.

Integration of Bass Enhancement and Active Noise Control System in Automobile Cabin

L Wang,W.S.Gan, S M Kuo
Journal Paper Special Issue on Active Noise Control, EURASIP Journal on Advances in Acoustics and Vibration, Vol. 2008, Article ID 869130, 9 pages, 2008.

Abstract

With the advancement of digital signal processing technologies, consumers are more concerned with the quality of multimedia entertainment in automobiles. In order to meet this demand, an audio enhancement system is needed to improve bass reproduction and cancel engine noise in the cabins. This paper presents an integrated active noise control system that is based on frequency-sampling filters to track and extract the bass information from the audio signal, and a multifrequency active noise equalizer to tune the low-frequency engine harmonics to enhance the bass reproduction. In the noise cancellation mode, a maximum of 3 dB bass enhancement can be achieved with significant noise suppression, while higher bass enhancement can be achieved in the bass enhance mode. The results show that the proposed system is effective for solving both the bass audio reproduction and the noise control problems in automobile cabins

Analysis of Misequalization in Narrowband Active Noise Equalizer System

L Wang,W.S.Gan
Journal Paper Journal of Sound and Vibration, Vol. 311, No. 3-5, pp 1438-1446, 8 April 2008.

Abstract

This paper presents the analysis of narrowband active noise equalizer (ANE) with imperfect secondary path estimation. Perfect secondary path estimation is often not available in practical cases. Under imperfect estimation condition, the system convergence and equalization exhibit properties not found in past literatures. The origin of such phenomenon is investigated. The maximum step size and misequalization are examined. Simulations were conducted to verify the theoretical analysis.

Active Noise Control System for Headphone Applications

Sen M. Kuo, Sohini MitraW.S.Gan
Journal Paper IEEE Transaction on Control Systems Technology, Vol. 14, No. 2, pp. 331-335, Mar 2006.

Abstract

This paper presents the design and implementation of an adaptive feedback active noise control (ANC) system for headphone applications. The ideal position of the error microphone in the ear-cup was studied and determined experimentally, and music signals were used for adaptive system identification of the secondary path. The designed ANC headphone was implemented using the TMS320C32 digital signal processor for real-time experiments. Performance has been evaluated and compared with a high-end commercial ANC headphone using the same set of primary noises including real-world engine noises. Experiment results show the proposed ANC headphone achieves higher noise cancellation, especially for low-frequency harmonics

Adaptive recurrent fuzzy neural networks for active noise control

Q Z Zhang, W.S.Gan
Journal Paper Journal of Sound and Vibration, Vol. 296, No. 4-5, pp 935-948, Oct 2006.

Abstract

This paper discussed nonlinear active noise control (ANC). Some adaptive nonlinear noise control approaches using recurrent fuzzy neural networks (RFNNs) were derived. The proposed RFNNs were feed-forward fuzzy neural networks (NNs) with different local feedback connections that are used to construct dynamic fuzzy rules. Different recurrent connection strategies, diagonal recurrent and full connected recurrent ones, were considered. In addition, different fuzzy operation strategies, product (multiply) inference and “summation” (addition) inference, were proposed. Because RFNN-based ANC systems can capture the dynamic behavior of a system through the feedback links, the exact lag of the input variables need not be known in advance. Online dynamic back-propagation learning algorithms based on the error gradient descent method were proposed, and the local convergence of a closed-loop system was proven using the discrete Lyapunov function. A nonlinear simulation example showed that an adaptive ANC system based on an RFNN with summation inference is superior to a system based on other fuzzy NNs.

Inherent Decorrelating and Least Perturbation Properties of the Normalized Subband Adaptive Filter

K A Lee, W.S.Gan
Journal Paper IEEE Transaction on Signal Processing, Vol. 54, No. 11, pp 4475-4480, Nov 2006.

Abstract

This correspondence describes and analyzes a class of subband adaptive filters (SAFs) that stems from various approaches of applying subband and multirate techniques in adaptive filtering. This class of SAFs, called the normalized SAF (NSAF), has a unique weight-control mechanism, whereby subband error signals are used to adapt a fullband tap-weight vector. In this correspondence, we elaborate on the inherent decorrelating properties of the NSAF, which manifest themselves in different forms as integral elements of the weight-control mechanism. We also show that the NSAF possesses some least perturbation properties that are also found in the normalized least-mean-square (NLMS) and affine projection (AP) algorithms

On the Subband Orthogonality of Cosine-Modulated Filter Banks

K A Lee, W.S.Gan
Journal Paper IEEE Transaction on Circuit and System (Part II), Vol. 53, No. 8, pp 677 – 681, Aug. 2006.

Abstract

In this brief, the second-order characteristics of cosine-modulated filter banks are formulated and analyzed. We show that, for any type of input spectrum, adjacent subband signals of a cosine-modulated filter bank are properly phase-aligned in such a way that they are nearly orthogonal at zero lag even though they are not mutually exclusive. The orthogonal properties are desirable in reducing the computational complexity of subband processing systems

Adaptive Feedback Active Noise Control Headset: Implementation, Evaluation and Its Extensions

W.S.Gan ,Sohini Mitra and Sen M. Kuo
Journal Paper IEEE Transaction on Consumer Electronics, Vol 51, No 3, pp 975-982, Aug 2005.

Abstract

In this paper, we present design and real-time implementation of a single-channel adaptive feedback active noise control (AFANC) headset for audio and communication applications. Several important design and implementation considerations, such as the ideal position of error microphone, training signal used, selection of adaptive algorithms and structures will be addressed in this paper. Real-time measurements and comparisons are also carried out with the latest commercial headset to evaluate its performance. In addition, several new extensions to the AFANC headset are described and evaluated.

Experimental Investigation of Active Vibration Control Using a Filtered-Error Neural Network and Piezoelectric Actuators

Y L Zhou, Q Z Zhang, X D Li,W.S.Gan
Journal Paper Lecture Notes in Computer Science, Vol. 3498, pp161-166, 2005.

Abstract

In this paper, we present design and real-time implementation of a single-channel adaptive feedback active noise control (AFANC) headset for audio and communication applications. Several important design and implementation considerations, such as the ideal position of error microphone, training signal used, selection of adaptive algorithms and structures will be addressed in this paper. Real-time measurements and comparisons are also carried out with the latest commercial headset to evaluate its performance. In addition, several new extensions to the AFANC headset are described and evaluated.

Analysis and DSP implementation of an ANC system using a filtered-error neural network

Y L Zhou, Q Z Zhang, X D Li, W.S.Gan
Journal Paper Journal of Sound and Vibration, Vol. 285, pp 1-25, 2005.

Abstract

In this paper, feedforward active noise control (ANC) using a neural network (NN) based on filtered-error back-propagation (BP) algorithm is considered. The filtered-error BP NN (FEBPNN) algorithm is first derived, and the difference between the FEBPNN algorithm and the filtered-X BP NN (FXBPNN) algorithm is given to show that the FEBPNN algorithm offers computational advantage over the FXBPNN algorithm. Computer simulations are carried out to compare the FEBPNN algorithm with the filtered-X least mean square (FXLMS) algorithm and the FXBPNN algorithm. The controllers based on the FEBPNN algorithm and the FXLMS algorithm are implemented on a Texas Instruments digital signal processor (DSP) TMS320VC33. The simulations and the experimental verification tests show that the FEBPNN algorithm performs as well as the FXLMS algorithm for a linear control problem, and better for a nonlinear control problem, at the same time, the simulations and the experimental verification tests also show that the convergence rate of the FEBPNN is acceptable, and the FEBPNN has better tracking ability under changes of the primal signal, the primary path and the secondary path. The experiments also lead to the conclusion that more work is required to improve the predictability and consistency of the performance of the NN controller based on the FEBPNN algorithm.

A Model Predictive Algorithm for Active Control of Nonlinear Noise Processes

Q Z Zhang, W.S.Gan, Y L Zhou
Journal Paper Journal of Shock and Vibration, Vol. 12, pp227-237, 2005.

Abstract

In this paper, an improved nonlinear Active Noise Control (ANC) system is achieved by introducing an appropriate secondary source. For ANC system to be successfully implemented, the nonlinearity of the primary path and time delay of the secondary path must be overcome. A nonlinear Model Predictive Control (MPC) strategy is introduced to deal with the time delay in the secondary path and the nonlinearity in the primary path of the ANC system. An overall online modeling technique is utilized for online secondary path and primary path estimation. The secondary path is estimated using an adaptive FIR filter, and the primary path is estimated using a Neural Network (NN). The two models are connected in parallel with the two paths. In this system, the mutual disturbances between the operation of the nonlinear ANC controller and modeling of the secondary can be greatly reduced. The coefficients of the adaptive FIR filter and weight vector of NN are adjusted online. Computer simulations are carried out to compare the proposed nonlinear MPC method with the nonlinear Filter-x Least Mean Square (FXLMS) algorithm. The results showed that the convergence speed of the proposed nonlinear MPC algorithm is faster than that of nonlinear FXLMS algorithm. For testing the robust performance of the proposed nonlinear ANC system, the sudden changes in the secondary path and primary path of the ANC system are considered. Results indicated that the proposed nonlinear ANC system can rapidly track the sudden changes in the acoustic paths of the nonlinear ANC system, and ensure the adaptive algorithm stable when the nonlinear ANC system is time variable.

Improving Convergence of the NLMS Algorithm Using Constrained Subband Updates

K A Lee, W.S.Gan
Journal Paper IEEE Signal Processing Letters, Vol. 11, No. 9, pp736-739, Sep 2004.

Abstract

We propose a new design criterion for subband adaptive filters (SAFs). The proposed multiple-constraint optimization criterion is based on the principle of minimal disturbance, where the multiple constraints are imposed on the updated subband filter outputs. Compared to the classical fullband least-mean-square (LMS) algorithm, the subband adaptive filtering algorithm derived from the proposed criterion exhibits faster convergence under colored excitation. Furthermore, the recursive tap-weight adaptation can be expressed in a simple form comparable to that of the normalized LMS (NLMS) algorithm. We also show that the proposed multiple-constraint optimization criterion is related to another known weighted criterion. The efficacy of the proposed criterion and algorithm are examined and validated via mathematical analysis and simulation.

Active Noise Control Using a Simplified Fuzzy Neural Network

Q Zhang,W.S.Gan
Journal Paper Journal of Sound and Vibration, Vol. 272, pp 437-449, Apr 2004.

Abstract

Active noise control (ANC) based on the principle of superposition has become an important and interesting topic of much research in recent years. In ANC system, a secondary source is introduced to generate anti-noise of equal amplitude and opposite phase with the primary noise.

A Model Predictive Algorithm for Active Noise Control with Online Secondary Path Modeling

Q Zhang,W.S.Gan
Journal Paper Journal of Sound and Vibration, Vol. 270, No.4-5, pp 1056-1066, Mar 2004

Abstract

In this paper, an improved nonlinear Active Noise Control (ANC) system is achieved by introducing an appropriate secondary source. For ANC system to be successfully implemented, the nonlinearity of the primary path and time delay of the secondary path must be overcome. A nonlinear Model Predictive Control (MPC) strategy is introduced to deal with the time delay in the secondary path and the nonlinearity in the primary path of the ANC system. An overall online modeling technique is utilized for online secondary path and primary path estimation. The secondary path is estimated using an adaptive FIR filter, and the primary path is estimated using a Neural Network (NN). The two models are connected in parallel with the two paths. In this system, the mutual disturbances between the operation of the nonlinear ANC controller and modeling of the secondary can be greatly reduced. The coefficients of the adaptive FIR filter and weight vector of NN are adjusted online. Computer simulations are carried out to compare the proposed nonlinear MPC method with the nonlinear Filter-x Least Mean Square (FXLMS) algorithm. The results showed that the convergence speed of the proposed nonlinear MPC algorithm is faster than that of nonlinear FXLMS algorithm. For testing the robust performance of the proposed nonlinear ANC system, the sudden changes in the secondary path and primary path of the ANC system are considered. Results indicated that the proposed nonlinear ANC system can rapidly track the sudden changes in the acoustic paths of the nonlinear ANC system, and ensure the adaptive algorithm stable when the nonlinear ANC system is time variable.

Applications of Adaptive Feedback active Noise Control System

S. M. Kuo, X Kong,W.S.Gan , A. Srinivasa
Journal Paper IEEE Trans on Control Systems Technology, Vol. 11, No. 2, 216-220, Mar 2003.

Abstract

This paper presents the experimental results of using the single-channel adaptive feedback active noise control (AFANC) algorithm with an innovative setup to achieve global attenuation of industrial machine noise in settings such as large manufacturing plants. An effective solution of using active/passive techniques and three distributed error sensors is proposed. The performance of the AFANC algorithm is verified by real-time experiments using the TMS320C32 DSP to control vibratory bowl and welding power generator noises. The experiments results show that this single-channel AFANC system can effectively reduce the noise level and is cost effective, portable, and easy for installation to control many noisy sources in large spaces.

An Integrated Audio Active Noise Control Headsets

W.S.Gan, S. M. Kuo
Journal Paper IEEE Transaction on Consumer Electronics, Vol.48, No.2, pp 242-247, May 2002.

Abstract

This paper presents an integrated approach in designing a noise reduction headset for audio and communication applications. Conventional passive headsets give good attenuation of ambient noise in the upper frequency range, while most of these devices fail below 500 Hz. Unlike the feedforward method, the adaptive feedback active noise control technique provides more accurate noise cancellation since the microphone is placed inside the ear-cup of the headset. Furthermore, the system uses a single microphone per ear cup, thus producing a more compact, lower power consumption, cheaper solution and ease of integration with existing audio and communication devices to form an integrated feedback active noise control headset. Simulation results have been conducted to show that the integrated approach can remove the disturbing noise and, at the same time, allow the desired speech or audio signal to pass through without cancellation

Robustness Analysis of a Environmental Active Noise Control System

J Yang, W.S.Gan, S E Tan
Journal Paper Journal of Sound and Vibration, Vol. 249, No. 3, pp 611-612, Jan 2002.

Abstract

On the Actively Controlled Noise Barrier

J Yang, W.S.Gan
Journal Paper Journal of Sound and Vibration, Vol. 240, No. 3, pp 592-597, 2001.

Abstract

Applying Equal-Loudness Compensation to the Adaptive Active Noise Control

W.S.Gan
Journal Paper Applied Acoustics, Vol. 61 No. 2, pp 183-187, Jun 2000.

Abstract

Previous works on the narrowband adaptive active noise control (AANC) have focused on using an adaptive gain factor to control the degree of noise reduction or amplification. This paper proposes a new method to incorporate equal-loudness compensation to the gain factor, thus allowing a more accurate perception of the actual attenuation or amplification of the noise level.

Broadband Active Noise Compressor

W Feng, W.S.Gan
Journal Paper IEEE Signal Processing Letter, USA, Vol. 5, No. 1, pp 11-14, Jan 1998.

Abstract

A broadband active noise compressor (ANCP) is presented to adapt the active noise equalization (ANE) technique suitable for practical usage. Compared to the existing broadband ANE system, the novel ANCP not only has the ability to shape the residual noise spectrum, but can also automatically adjust the residual noise power. This algorithm is analyzed in steady state and verified by computer simulations.

Designing the fuzzy step-size LMS algorithm

W.S.Gan
Journal Paper IEE Proceedings- Vision, Image and Signal Processing, Vol. 144, Issue 5, pp 261-266, Oct 1997.

Abstract

A new approach in adjusting the step size of the least mean square (LMS) using the fuzzy logic technique is presented. It extends the earlier work of Gan (see Signal Process., vol.49, no.2, p.145-49, 1996) by giving a complete design methodology and guidelines for developing a reliable and robust fuzzy step size LMS (FSS LMS) algorithm. It also presents a computational study and simulation results of this newly proposed algorithm compared to other conventional variable step size LMS algorithms