A new two‐microphone Gauss‐Seidel pseudo affine projection algorithm for speech quality enhancement

This study addresses the problem of speech quality enhancement by adaptive and nonadaptive filtering algorithms. The well‐known two‐microphone forward blind source separation (TM‐FBSS) structure has been largely studied in the literature. Several two‐microphone algorithms combined with TM‐FBSS have been recently proposed. In this study, we propose 2 contributions: In the first, a new two‐microphone Gauss‐Seidel pseudo affine projection (TM‐GSPAP) algorithm is combined with TM‐FBSS. In the second, we propose to use the new TM‐GSPAP algorithm in speech enhancement. Furthermore, we show the efficiency of the proposed TM‐GSPAP algorithm in speech enhancement when highly noisy observations are available. To validate the good performances of our algorithm, we have evaluated the adaptive filtering properties in computational complexity and convergence speed performance by system mismatch criteria. A fair comparison with adaptive and nonadaptive noise reduction algorithms are also presented. The adaptive algorithms are the well‐known two‐microphone normalized least mean square algorithm, and the recently published two‐microphone pseudo affine projection algorithm. The nonadaptive algorithms are the one‐microphone spectral subtraction and the two‐microphone Wiener filter algorithm. We evalute the quality of the output speech signal in each algorithm by several objective and subjective criteria as the segmental signal‐to‐noise ratio, cepstral distance, perceptual evaluation of speech quality, and the mean opinion score. Finally, we validate the superior performances of the proposed algorithm with physically measured signals.     

Real‐valued MUSIC algorithm for power harmonics and interharmonics estimation

Recently, advanced spectrum estimation methods, including the MUSIC (Multiple Signal Classification) algorithms, are being gradually employed for high‐resolution power harmonics analysis. However, most of them are proposed to detect frequencies of complex‐valued signals, so that any real‐valued signal should be transformed into complex form. This data pre‐treatment may lead to additional computation burden. In addition, the picket‐fence effects also exist as in the FFT algorithm and cause poor frequency resolution. To overcome these drawbacks, a real‐valued MUSIC algorithm is proposed for power harmonics analysis in this paper. The algorithm is based on the subspace decomposition theory and the computation of pseudospectrum is also provided. Additionally, to improve the measuring precision, the Newton–Raphson algorithm is adopted to optimize the harmonic frequencies significantly. Simulation results show that, in the real‐valued MUSIC pseudospectrum, the spectral peaks of actual harmonic components can be more easily distinguished from the false peaks caused by noise, and the computational complexity is notably lower than that of the classic complex MUSIC, as well as the detecting accuracy is close to that of root‐MUSIC algorithm which is quite time consuming. Experimental results prove that the proposed strategy is more suitable for high‐resolution power harmonics estimation. Copyright © 2010 John Wiley & Sons, Ltd.     

A new fast Newton‐type adaptive filtering algorithm for stereophonic acoustic echo cancellation (SAEC)

This paper addresses the problem of acoustic echo cancellation. We propose a new version of the fast Newton transversal filter algorithm for stereophonic acoustic echo cancellation applications. This algorithm can be viewed as an efficient implementation of the extended two‐channel fast transversal filter algorithm. Moreover, it fits naturally within the frame of the fast version of the recursive least‐squares (RLS) algorithm, applied to the two‐channel case. To stabilize the proposed two‐channel algorithm, we have adapted and then applied a new numerical stabilization technique that has been proposed recently. The computational complexity of the proposed two‐channel algorithm is less than half the complexity of the fastest two‐channel RLS versions and very close to that of the two‐channel normalized least mean squares algorithm when its predicting part length is chosen to be small. Simulation results and comparisons in term of complexities, convergence speed and tracking with the two‐channel algorithms are presented. Copyright © 2009 John Wiley & Sons, Ltd.     

Zero‐attracting variable‐step‐size least mean square algorithms for adaptive sparse channel estimation

Recently, sparsity‐aware least mean square (LMS) algorithms have been proposed to improve the performance of the standard LMS algorithm for various sparse signals, such as the well‐known zero‐attracting LMS (ZA‐LMS) algorithm and its reweighted ZA‐LMS (RZA‐LMS) algorithm. To utilize the sparsity of the channels in wireless communication and one of the inherent advantages of the RZA‐LMS algorithm, we propose an adaptive reweighted zero‐attracting sigmoid functioned variable‐step‐size LMS (ARZA‐SVSS‐LMS) algorithm by the use of variable‐step‐size techniques and parameter adjustment method. As a result, the proposed ARZA‐SVSS‐LMS algorithm can achieve faster convergence speed and better steady‐state performance, which are verified in a sparse channel and compared with those of other popular LMS algorithms. The simulation results show that the proposed ARZA‐SVSS‐LMS algorithm outperforms the standard LMS algorithm and the previously proposed sparsity‐aware algorithms for dealing with sparse signals. Copyright © 2015 John Wiley & Sons, Ltd.     

Complex two‐dimensional TNIPM for l1 norm‐based sparse optimization to collocated MIMO radar

In collocated multiple‐input multiple‐output (MIMO) radar, because of the sparse nature of the received signal in the three dimensions of range, angle, and Doppler, accurate estimates of range/angle/Doppler parameters can be achieved using a sparse signal recovery. In this paper, we develop a complex two‐dimensional truncated Newton interior point method (2D TNIPM) for l₁‐norm‐based sparse optimization. Because of the 2D sparse representation of received signal in collocated MIMO radar systems, the performance of proposed algorithm is investigated in order to estimate the target position and velocity. Simulation results show that the 2D TNIPM requires much lower computations compared to the 1D one. Also, it outperforms some other 2D algorithms in the estimation of range, angle, and Doppler parameters under low signal‐to‐noise ratios. © 2015 Institute of Electrical Engineers of Japan. Published by John Wiley & Sons, Inc.     

Filtering,smoothing and prediction using a control‐loop spectral factorization method for coloured noise

A method for the linear least‐squares estimation of random signals contaminated with random noise that uses a new method of spectral factorization is shown. It is shown that the optimal filter can be written entirely in terms of the two spectral factors of signal plus noise and noise‐alone, and can be applied to the general case of coloured and white additive noise. The method of spectral factorization used is novel and uses control‐system methodology. Copyright © 2012 John Wiley & Sons, Ltd.     

Blind adaptive identification of 2‐channel systems using bias‐compensated RLS algorithm

This paper studies the problem of blind adaptive identification, which focuses on how to obtain the consistent estimation of channel characteristics when only the output signal of each transmission channel is available. To solve this problem, traditional algorithms usually construct a single‐input–multiple‐output system resorting to the technique of antenna array or time oversampling. However, they simply suppose that the noise of each channel is known a priori or balanced, which cannot always be satisfied in practice. Therefore, considering the practical situation where the noise of each transmission channel is both unknown and unbalanced, a bias‐compensated recursive least‐squares algorithm is proposed, which can estimate the unbalanced noises in real time and obtain the consistent estimation of channel characteristics. Simulation results illustrate the good performance of the proposed algorithm under different signal‐to‐noise‐ratio conditions.     

90dB‐DR 3.3V CMOS single‐ended‐to‐fully differential and fully differential‐to‐single‐ended amplifiers for audio applications

This paper presents the design and the realization of single‐ended‐to‐fully differential and fully differential‐to‐single‐ended amplifiers to be used in an audio signal processing system. The proposed blocks allow to reduce significantly the pin number of the developed system, while guaranteeing the high quality (16bit) performance required in an audio channel. The proposed circuits have been realized in a standard 3.3V 0.35 µm CMOS technology and achieve a Dynamic Range in excess of 90dB with a Total Harmonic Distortion lower than ‐80dB for a full scale signal amplitude. Their power consumption (≈6mW and each) and the area (0.1mm² each) are finally negligible with respect to the other blocks in the overall systems. Copyright © 2003 John Wiley & Sons, Ltd.     

Signal separation using second‐ and high‐order statistics

This paper presents two methods for signal separation. In either method, the fundamental criterion for separation relies on reducing to zero, or at least minimizing, the output cross‐correlation or cross‐cumulant functions of a decoupling multi‐input–multi‐output system that is fed with mixed signals. In one of the approaches used, the parameters of this system are determined through solving — in a least‐squares sense — a linearized set of equations describing the deviations from zero of either the cross‐correlation or cross‐cumulant functions when evaluated for different lags. An alternative rapidly convergent adaptive algorithm is also described for minimizing the cross‐correlation or cross‐cumulant functions. The paper also considers both FIR and IIR representations of the decoupling system. It shows that using IIR functions in the decoupling system does not offer any merit over the FIR case. Illustrative examples are given to show the performance of the proposed algorithms. Copyright © 2000 John Wiley & Sons, Ltd.     

An efficient adaptive algorithm for edge detection based on the likelihood ratio test

The edge detection problem in blurred and noisy 2‐D signals is dealt with. An adaptive signal processing algorithm is proposed which marks edge points according to an hypothesis test which compares the likelihoods of two models describing the local signal behaviour in the two cases of absence/presence of an edge. The two models are identified by a regularized least squares estimation algorithm, obtaining a numerically efficient procedure, quite robust with respect to additive noise and blurr perturbation. No global thresholding or data prefiltering is required. Copyright © 2002 John Wiley & Sons, Ltd.     

基-2 FFT在音频分析及输出系统中的应用

主要介绍了一种将FFT算法移植到单片机上运行,通过对数字音频信号进行分析处理,以实现音乐频谱实时显示和声音输出的系统。系统硬件部分主要由声音输入、单片机模数转换、由LED组成的点阵单元以及声音信号放大输出等电路组成。利用高性能STC12C5A60S2单片机内建的模数转换功能,先将输入的音频信号采样、量化转换为数字信号,再通过软件编程进行FFT运算。输出处理结果点亮LED点阵,完成频谱显示。LED的明暗由音乐的频率变化决定。通过LM386运放芯片及外围电路将输入的音频信号进行放大后,由喇叭或者外接音箱输出。该设计不但具有较高的实用价值和观赏性,而且硬件电路结构简洁,开发、制作成本低。     

Novel design for non‐latency wireless binaural hearing aids

Binaural hearing aids consist of two hearing devices, one for each ear. A new concept of binaural hearing aids is proposed, in which only the master hearing aid contains a Bluetooth chip for receiving stereo audio signals from an external device, and the signal in one channel is sent to the slave hearing aid from the master by a 2.4‐GHz Gaussian frequency‐shift keying (GFSK) RF transmission method to create the binaural hearing effect. However, a problem arises in regard to the processing necessary for the signal transmission and reception in the two hearing aids, which creates a time delay that causes the precedence effect. Therefore, an audio delay processing algorithm has been designed in the master hearing aid to synchronize with the sound output of the slave hearing aid. Experimental results show that the time difference between the two hearing aids is about 8 µs, which is effective for avoiding the precedence effect. © 2014 Institute of Electrical Engineers of Japan. Published by John Wiley & Sons, Inc.     

A fast second‐order signal separation algorithms with on‐line capabilities

In correlation‐based signal separation algorithms, the received mixed signals are fed to a de‐coupling system designed to minimize the output cross‐correlation functions. If minimizaion is perfect, each of the system's outputs carries only one signal independent of the others. In these algorithms, the computation burden of the output cross‐correlation functions normally slows down the separation algorithm. This paper, describes a computationally efficient method for off‐line pre‐computation of the needed cross‐correlation functions. Explicit formulas have been derived for the output cross‐correlation functions in terms of the received input signals and the de‐coupling system parameters. Then, it is shown that signal separation amounts to the least‐squares solution of a system of linear equations describing these output cross‐correlation functions, evaluated over a batch of lags. Next, a fast RLS‐type adaptive algorithm is devised for on‐line signal separation. In this respect, an algorithm is derived for updating the de‐coupling parameters as data comes in. This update is achieved recursively, along the negative of the steepest descent directions of an objective cost function describing the output cross‐correlation functions over a batch of lags, subject to equal output power constraints. Illustrative examples are given to demonstrate the effectiveness of the proposed algorithms. Copyright © 2002 John Wiley & Sons, Ltd.     

电力线通信系统中的信道训练和信道估计方案

在对电力线信道进行估计的过程中采用信号处理方法,将通信数据与部分已知的训练数据相结合,通过在频域和时域进行信号变换和算法处理实现了电力线通信中的信道训练与数据通信的兼容性和实时性,节省了信道的带宽并提高了信道估计的准确性和稳定性。最后通过实验证明了所提出的信号处理算法的有效性。     

On the stability and convergence of a sliding‐window variable‐regularization recursive‐least‐squares algorithm

A sliding‐window variable‐regularization recursive‐least‐squares algorithm is derived, and its convergence properties, computational complexity, and numerical stability are analyzed. The algorithm operates on a finite data window and allows for time‐varying regularization in the weighting and the difference between estimates. Numerical examples are provided to compare the performance of this technique with the least mean squares and affine projection algorithms. Copyright © 2015 John Wiley & Sons, Ltd.     

Development of numerical linear algebra algorithms in dynamic fixed‐point format: a case study of Lanczos tridiagonalization

Proper range and precision analysis play an important role in the development of fixed‐point algorithms for embedded system applications. Numerical linear algebra algorithms used to find singular value decomposition of symmetric matrices are suitable for signal and image‐processing applications. These algorithms have not been attempted much in fixed‐point arithmetic. The reason is wide dynamic range of data and vulnerability of the algorithms to round‐off errors. For any real‐time application, the range of the input matrix may change frequently. This poses difficulty for constant and variable fixed‐point formats to decide on integer wordlengths during float‐to‐fixed conversion process because these formats involve determination of integer wordlengths before the compilation of the program. Thus, these formats may not guarantee to avoid overflow for all ranges of input matrices. To circumvent this problem, a novel dynamic fixed‐point format has been proposed to compute integer wordlengths adaptively during runtime. Lanczos algorithm with partial orthogonalization, which is a tridiagonalization step in computation of singular value decomposition of symmetric matrices, has been taken up as a case study. The fixed‐point Lanczos algorithm is tested for matrices with different dimensions and condition numbers along with image covariance matrix. The accuracy of fixed‐point Lanczos algorithm in three different formats has been compared on the basis of signal‐to‐quantization‐noise‐ratio, number of accurate fractional bits, orthogonality and factorization errors. Results show that dynamic fixed‐point format either outperforms or performs on par with constant and variable formats. Determination of fractional wordlengths requires minimization of hardware cost subject to accuracy constraint. In this context, we propose an analytical framework for deriving mean‐square‐error or quantization noise power among Lanczos vectors, which can serve as an accuracy constraint for wordlength optimization. Error is found to propagate through different arithmetic operations and finally accumulate in the last Lanczos vector. It is observed that variable and dynamic fixed‐point formats produce vectors with lesser round‐off error than constant format. All the three fixed‐point formats of Lanczos algorithm have been synthesized on Virtex 7 field‐programmable gate array using Vivado high‐level synthesis design tool. A comparative study of resource usage and power consumption is carried out. Copyright © 2015 John Wiley & Sons, Ltd.     

New efficient adaptive fast transversal filtering (FTF)‐type algorithms for mono and stereophonic acoustic echo cancelation

This paper addresses the field of stereophonic acoustic echo cancelation (SAEC) by adaptive filtering algorithms. Recently, simplified versions of the fast transversal filter (SFTF)‐type algorithm has been proposed. In this paper, we propose two major contributions. In the first contribution, we propose two new FTF‐type algorithms with low complexity and good convergence speed characteristics. These two proposed algorithms are mainly on the basis of a forward prediction scheme to estimate the so called dual Kalman gain, which is inherent in the filtering part update. This computation complexity is achieved by the introduction of new relations for the computation of the likelihood variables that are simple and lead to further simplifications on the prediction part of the two proposed algorithms. In the second contribution, we propose to adapt then apply these four new SFTF‐type algorithms, (the two proposed algorithms in this paper and their original versions) in the SAEC applications. A fair comparison of the proposed algorithms with the original SFTF and the normalized least mean square algorithms, in mono and SAEC applications, is presented. Copyright © 2014 John Wiley & Sons, Ltd.     

基于图信号处理的智能电表功率信号分解

智能电表的大规模部署,使得对电表采集的低频信号进行数据分析成为一个研究热点。以非侵入式负荷监测为背景,研究基于图信号处理(GSP)的低频功率信号分解算法。首先,将功率信号分解定义为最小化求解问题,并引入基于图转移矩阵的全局变化量作为正则项。然后,分两步对该优化问题求解:第1步最小化正则项得到满足图信号全局变化量最小的近似解;第2步以该解为基础,利用模拟退火算法对目标函数和约束条件迭代寻优。最后利用开源数据库REDD进行仿真,验证了该算法在分类准确率上的优势,且与其他算法相比对训练数据的依赖性较小。     

Development of 3‐D equivalent‐circuit modelling with decoupled L‐ILU factorization in semiconductor‐device simulation

In this paper, we develop a three‐dimensional (3‐D) device simulator, which combines a simplified, decoupled Gummel‐like method equivalent‐circuit model (DM) with levelized incomplete LU (L‐ILU) factorization. These complementary techniques are successfully combined to yield an efficient and robust method for semiconductor‐device simulation. The memory requirements are reduced significantly compared to the conventionally used Newton‐like method. Furthermore, the complex voltage‐controlled current source (VCCS) is simplified as a nonlinear resistor. Hence, the programming and debugging for the nonlinear resistor model is much easier than that for the VCCS model. Further, we create a connection‐table to arrange the scattered non‐zero fill‐ins in sparse matrix to increase the efficiency of L‐ILU factorization. Low memory requirements may pave the way for the widespread application in 3‐D semiconductor‐device simulation. We use the body‐tied silicon‐on‐insulator MOSFET structure to illustrate the capability and the efficiency of the 3‐D DM equivalent‐circuit model with L‐ILU factorization. Copyright © 2007 John Wiley & Sons, Ltd.     

Generation of band‐pass constant‐envelope signals with a pre‐assigned spectrum: a synthesis procedure

The recent, increasing adoption of spread‐spectrum techniques, either in communications or in other fields, makes the design of band‐pass sources a topical subject. We consider the generation of constant‐envelope (i.e. constant power) band‐pass signals via a frequency modulation technique. In particular, we propose a general solution to the synthesis problem of delivering a constant envelope signal with a pre‐assigned band‐pass spectrum. Both quasi‐stationary and non‐stationary modulations are discussed and a novel optimization algorithm is proposed for the latter case. A design example concludes the presentation. Copyright © 2002 John Wiley & Sons, Ltd.