Jun Yang

An Improved Multiband-Structured Subband Adaptive Filter Algorithm

Recently, a multiband-structured subband adaptive filter (MSAF) algorithm was proposed to speed up the convergence of the normalized least-mean-square (NLMS) algorithm. In this letter, we extend this work and propose an improved multiband-structured subband adaptive filter (IMSAF) algorithm to increase the convergence speed of the MSAF, which can also be regarded as a unifying framework for the NLMS, MSAF, and affine projection (AP) algorithms. The proposed optimization criterion is based on the principle of minimal disturbance, canceling the most recent P a posteriori errors in each of the N subbands. The stability condition and the computational complexity are also analyzed. Computer simulations in the context of system identification demonstrate the effectiveness of the new algorithm.

ADAPTIVE DETECTION OF MULTIPLE POINT-LIKE TARGETS UNDER CONIC CONSTRAINTS

This paper addresses the problem of detecting multiple point-like targets in the presence of steering vector mismatches and Gaussian disturbance with unknown covariance matrix. To this end, we flrst model the actual useful signal as a vector belonging to a proper cone whose axis coincides with the whitened direction of the nominal array response. Then we develop two robust adaptive detectors resort- ing to the two-step GLRT-based design procedure without assignment of a distinct set of secondary data. The performance assessment has been conducted by Monte Carlo simulation, also in comparison to pre- viously proposed detectors, and conflrms the efiectiveness of the newly proposed ones. In the last part of the work, in order to restore the detection performance of the newly proposed detectors in the presence of a large number of range cells contaminated by useful signals, we consider two adaptive detectors which resort to the structure informa- tion of the disturbance covariance matrix, and show that the a-priori information on the covariance structure can lead to a noticeable per- formance improvement.

Steady-State Solution of the Deficient Length Constrained FBLMS Algorithm

In almost all analyses of the frequency-domain block least mean-square (FBLMS) algorithm, the length of the adaptive filter is assumed to be equal to that of the unknown system impulse response. However, in many practical situations, where the length of the system impulse response is extremely long, the adaptive filter usually works in an under-modeling situation. The analysis results for the sufficient length FBLMS algorithm are not applicable to the deficient length case. In this correspondence, the steady-state solution of the deficient length constrained FBLMS algorithm is analyzed. Simulations illustrate the accuracy of the theoretical results.

A fast exact filtering approach to a family of affine projection-type algorithms

The affine projection (AP)-type algorithms produce a good tradeoff between convergence speed and complexity. As the projection order increases, the convergence rate of the AP algorithm is improved at a relatively high complexity. Many efforts have been made to reduce the complexity. However, most of the efficient versions of the AP-type algorithms are based on the fast approximate filtering (FAF) scheme originally proposed in the fast AP (FAP) algorithm. The approximation leads to degraded convergence performance. Recently, a fast exact filtering (FEF) AP (FEAP) algorithm was proposed by Y. Zakharov. In this paper, we propose a new FEF approach to further reduce the complexity of the FEAP algorithm given that the calculation of the weight vector is not the primary objective for the application at hand. The proposed FEF scheme is then extended to the dichotomous coordinate descent (DCD)-AP, affine projection sign (APS), and modified filtered-x affine projection (MFxAP) algorithms. The complexity of AP-type algorithms based on the proposed FEF approach is comparable to that based on the FAF scheme. Moreover, analysis results show that the complexity reduction of the new algorithms is achieved without any performance degradation. HighlightsWe propose a new fast exact filtering approach to a family of AP-type algorithms.The error vector is calculated using an auxiliary coefficient vector instead of a true weight vector.The complexity of AP-type algorithms based on the proposed fast exact filtering approach is comparable to that based on the fast approximate filtering scheme.

Transient and steady-state analyses of the improved multiband-structured subband adaptive filter algorithm

The authors previously proposed an improved multiband-structured subband adaptive filter (IMSAF) algorithm. In this contribution, they first present two delayless structures of the IMSAF algorithm to remove delay in the signal path. Then, they study the transient and steady-state behaviour of the IMSAF algorithms based on the energy conservation arguments and paraunitary condition imposed on the analysis and synthesis filter banks. The analysis does not require a model for the input signal. Simulation results show that the proposed delayless IMSAF algorithm has a faster convergence rate than the traditional delayless subband adaptive filtering algorithms. Theoretical analysis of the IMSAF algorithm is in good agreement with computer simulation results.

Development of an Acoustic Filter for Parametric Loudspeaker in Air

In this study, an acoustic filter for a parametric loudspeaker for removing the spurious signal generated by the nonlinearity of the receiving transducer was developed. A 16-mm-diameter aluminum plate was set in front of the transducer to attenuate ultrasound by more than 15 dB on average, with negligible effect on the audible sound. Experimental results demonstrated that, in the near field of the parametric loudspeaker, the spurious signal was dominant in the measurements as the sound pressure levels of the primary waves were more than 120 dB. However, the audible sound generated by the parametric loudspeaker can be detected even without the acoustic filter in the far field.

Low-Complexity Implementation of the Improved Multiband-Structured Subband Adaptive Filter Algorithm

Previously, we proposed an improved multiband-structured subband adaptive filter (IMSAF) algorithm to accelerate the convergence rate of the MSAF algorithm. When the projection order and/or the number of subbands is increased, the convergence rate of the IMSAF algorithm improves at the cost of increased complexity. Thus, this paper proposes several approaches to reduce the complexity of the IMSAF algorithm, both in error vector calculation and matrix inversion operation. Specifically, three approaches are developed to efficiently calculate error vector. The first approach gives an approximate filtering, whereas the other two approaches can provide a fast exact filtering with or without update of the weight vector explicitly based on a recursive scheme. The decorrelation property of IMSAF is determined, and two simplified variants are developed to reduce the complexity as by-products, i.e., the simplified IMSAF (SIMSAF) and pseudo IMSAF algorithms. Then, we discuss the problem of solving a linear system of equations. The performance advantages, limitations, and preferable applications of various methods are analyzed and discussed. Computer simulations are conducted in the context of system identification to determine the principle and efficiency of the proposed fast algorithms.

Generation of Audible Sound from Two Ultrasonic Beams with Dummy Head

In this study, the nonlinear interaction of two sound beams with a Knowles Electronics manikin for acoustic research (KEMAR) dummy head was investigated experimentally. An audible sound was generated under two conditions: two ultrasonic beams were in parallel and at right angle. It was found that the presence of the dummy head not only increased audible sound levels in KEMAR ears owing to a head scattering effect, but also led to different sound levels in two ears owing to a head shadow effect. Experimental results demonstrated that for the two parallel ultrasonic beams, head effects slightly reduced the audible sound levels behind the dummy head, but increased the audible sound area. Furthermore, it was shown that the audible sound could be generated within the intersection zone of two perpendicular ultrasonic beams, which is useful for localized sound delivery in audio engineering.

Wireless communication using ultrasound through metal barriers: Experiment and analysis

Many industrial applications need transmission of information through metal barriers without physical penetrations. Conventional wireless communication is ineffective due to the Faraday shielding effect. Ultrasound can be used to convey information in this situation. To achieve lower power consumption and higher data rate simultaneously, an ultrasonic communication system based on single-carrier frequency domain equalization (SC-FDE) has been proposed. Compared with conventional ultrasonic communication systems based on orthogonal frequency division multiplexing (OFDM), it has a lower peak-to-average power ratio while maintaining a similar performance in anti-multipath fading. Unique word sequences are used to realize the frame synchronization and to estimate the communication channel characteristics. A prototype system has been implemented on a field programmable gate array and a digital signal processor. It gets an effective bit rate of 436 kbps through a steel barrier within the thickness of 70 mm, and higher data rate will be achieved by using higher frequency channels or higher order modulation methods. Due to the low-complexity merit of the transmitter, the proposed system could work for a long time even under battery-powered conditions, which makes it practical for many industrial applications.

Fast implementation of a family of memory proportionate affine projection algorithm

Previously, a family of memory proportionate affine projection (MPAP) algorithms has been proposed by taking into account the “history” of the proportionate factors for sparse system identification. This paper presents a low-complexity implementation of this family of MPAP algorithms. Two most important ideas are used in the derivation of the fast algorithm. The first one is to update the auxiliary coefficient vector rather than the true coefficient vector. The second interesting idea is to calculate the error vector by using a recursive technique. Simulation results demonstrate the effectiveness of the fast algorithms.