Woon-Seng Gan

Improving convergence of the NLMS algorithm using constrained subband updates

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 system for headphone applications

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

Recent advances on active noise control : open issues and innovative applications

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.

A broadband self-tuning active noise equaliser

Abstract This paper presents a broadband Self-tuning Active Noise Equaliser (SANE) which improves the existing ANE system to meet the practical requirements. By introducing a variable gain factor, the novel SANE has two functions: it can not only shape the residual noise spectrum as the existing ANE does, but it can automatically adjust the residual noise power as well. This algorithm is supported by computer simulations.

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

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 kappa-space domain by applying Fourier transform. The kappa-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 Ocheltrees 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.

A digital beamsteerer for difference frequency in a parametric array

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 (/spl sim/26/spl deg/) 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 frequencys beamwidth and sidelobe. Most importantly, the proposed system can steer the difference frequency to a small angle with minimal computation.

Efficient VLSI Architecture for Lifting-Based Discrete Wavelet Packet Transform

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

An integrated environment for rapid prototyping of DSP Algorithms using matlab and Texas instruments’ TMS320C30

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.

Teaching DSP software development: from design to fixed-point implementations

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.

Modeling of finite-amplitude sound beams: second order fields generated by a parametric loudspeaker

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.