Subband adaptive filtering with real-valued subband signals foracoustic echo cancellation

Adaptive filtering in subbands was originally proposed to overcome the limitations of conventional least-mean-square (LMS) algorithms. In general, subband adaptive filters offer computational savings, as well as faster convergence over the conventional LMS algorithm. However, improvements to current subband adaptive filters could be further enhanced by a more elegant choice of their design/structure. Classical subband adaptive filters employ DFT-based analysis and synthesis filter banks which results in subband signals that are complex-valued. The authors modify the structure of subband adaptive filters by using single-sideband (SSB) modulated analysis and synthesis filter banks, which result in subband signals that are real-valued. This simplifies the realisation of subband adaptive filters     

A subband adaptive filter allowing maximally decimation

Conventional subband adaptive filters (ADF) using filter banks have shown degradation in performance because of the nonideal nature of filter banks. For this problem, the authors propose an alias free subband structure for adaptive filtering using polyphase frequency sampling filter (FSF) banks. As a preliminary, they make it clear that the conventional polyphase discrete Fourier transform (DFT) bank is equivalent to a class of the FSF bank. Then, they propose a new class of ADF using the FSF banks based on the frequency sampling theorem. As a result, the proposed technique enables subband adaptive filtering without the degradation effect of both the aliasing and cross terms, even if one chooses critical subsampling     

Convergence analysis of alias-free subband adaptive filters based on a frequency domain technique

Convergence analysis of alias-free subband adaptive filters (SADFs) is presented based on a frequency-domain technique where instead of analyzing the adaptive algorithms in the time-domain, the averaging method and the ordinary differential equation (ODE) method are applied to the frequency-domain expressions of the adaptive algorithms converted by the discrete Fourier transform. As an alias-free SADF algorithm, the SADF proposed by Pradhan and Reddy is known. In this paper, this technique is first applied to the Pradhan's SADF. The stability of the Pradhan's SADF is verified in the frequency domain, and a simple formula to evaluate the mean square error (MSE) of the algorithm is theoretically derived. By using a slight modification, the technique can be applied to the two-band delayless subband adaptive filter (DLSADF) with the Hadamard transform. The stability condition and the MSE of the DLSADF with the Hadamard transform are also obtained. Simulation results of both algorithms show the validity of the theoretical results.     

Adaptive array processing for multipath fading mitigation viaexploitation of filter banks

We propose subband adaptive array processing for mitigation of both intersymbol interference (ISI) and cochannel interference (CCI) in digital mobile communications. Subband adaptive array processing employs filter banks in a front end to an adaptive array receiver. By decomposing the signals into a set of subband signals, the analysis filters enhance the correlation of multipath rays in each subband. This enhancement is blind in the sense that no a priori knowledge of the temporal characteristics or spatial signatures of arriving signals is required. With the increased coherence, the desired signal can be effectively equalized by subsequent spatial processing. Further, the CCI signals and their multipaths can be suppressed with fewer degrees of freedom. The effects of quadrature mirror filter and discrete Fourier transform filter banks on multipath correlation are delineated     

基于二维APDCSF的列率子带特征编码方法

提出了一种子带编码的新方法。该方法利用二维全相位离散反余弦列率滤波器（APDCSF）对图像进行子带分解;对于低频子带图像采用直接斜交多重亚采样和基于全相位离散反余弦列牢滤波器（APDICsF）的多重旋转内插恢复．而对高频子带图像利用直方图自动阈值化提取如边缘和线等特征的图像元;根据各个子带的图像元的特征分别进行编码压缩,解压缩后利用凸集投影重建原始图像。该方法消除了传统的离散余弦变换（DCT）编码的方块化效应,与基于小波变换的子带特征编码方法相比,计算复杂度小,压缩率高,主观视觉性能好,对于灰阶图像可达到0．1～0．3bpp,特别适用于低比特率图像压缩。     

一种二维离散子波变换的滤波器结构

子波变换具有良好的时间（空间）频率局部化性能，在图象子带编码中二维离散子波变换是一种接近理想的子带分析／综合子系统．本文提出一种利用一维离散子波变换实现二维有限长离散子波变换的方法，同时给出了二维离散子波正变换（DWT）和反变换（IDWT）的滤波器实现结构．实验结果表明新的方法具有良好的重构性，完全适用于图象压缩编码系统中的分析／综合子系统．     

Efficient design of oversampled NPR GDFT filterbanks

We propose a flexible, efficient design technique for the prototype filter of an oversampled near perfect reconstruction (NPR) generalized discrete Fourier transform (GDFT) filterbank. Such filterbanks have several desirable properties for subband processing systems that are sensitive to aliasing, such as subband adaptive filters. The design criteria for the prototype filter are explicit bounds (derived herein) on the aliased components in the subbands and the output, the distortion induced by the filterbank, and the imaged subband errors in the output. It is shown that the design of an optimal prototype filter can be transformed into a convex optimization problem, which can be efficiently solved. The proposed design technique provides an efficient and effective tool for exploring many of the inherent tradeoffs in the design of the prototype filter, including the tradeoff between aliasing in the subbands and the distortion induced by the filterbank. We calculate several examples of these tradeoffs and demonstrate that the proposed method can generate filters with significantly better performance than filters obtained using current design methods.     

Discrete Sine and Cosine Transforms in Single Carrier Modulation Systems

In the DFT-based single carrier frequency division multiple access (SC-FDMA) modulation system, the discrete Fourier transform (DFT) is usually exploited to divide the frequency channel. In this paper, we propose to use discrete sine/cosine transforms in place of the DFT for SC-FDMA. Eight SC-FDMA systems based on various DST/DCT types are studied. The bit error rate (BER) and the peak to average power ratio (PAPR) are used to evaluate performance of these eight DST/DCT-based SC-FDMA systems in comparison with the conventional DFT-based SC-FDMA system in an AWGN environment. Simulation results show that the DST/DCT-based SC-FDMA systems with the use of localized FDMA scheme can provide better BER performance and yet keep the same PAPR performance as compared to the DFT-based SC-FDMA system.     

非平稳海杂波背景下子带开关相干检测器

海杂波由纹理分量调制散斑分量构成,本文着重研究纹理分量和散斑分量的平稳特性.首先,通过DFT调制滤波器组将海杂波分解到各个子带,提出两种一致性因子分别衡量各个子带海杂波的纹理分量和散斑分量的平稳性,实验结果表明海杂波的非平稳特性主要体现在纹理分量上,强杂波区的纹理分量体现出显著的非平稳性,弱杂波区的纹理分量具有平稳特性.进一步针对海杂波在不同多普勒频率分量上具有不同的平稳特性,将纹理分量的一致性因子作为判断函数,分别使用样本协方差矩阵和归一化样本协方差矩阵估计算法,提出了子带开关自适应归一化匹配滤波(Adaptive Normalized Matched Filter,ANMF)检测器.实测的海杂波数据实验结果表明,子带开关ANMF检测器的性能优于对比算法.     

Delayless subband adaptive filtering using the Hadamard transform

In this correspondence, an analysis of a delayless critically decimated subband adaptive filter structure is presented. In this structure, adaptive weights in each subband are computed by the LMS algorithm and then transformed into those in fullband by the Hadamard transform. It is shown that a stationary point of the proposed algorithm corresponds to the fullband Wiener filter. Some numerical results are also presented to show the performance of this scheme     

On the modeling of DCT and subband image data for compression

Image subband and discrete cosine transform coefficients are modeled for efficient quantization and noiseless coding. Quantizers and codes are selected based on Laplacian, fixed generalized Gaussian, and adaptive generalized Gaussian models. The quantizers and codes based on the adaptive generalized Gaussian models are always superior in mean-squared error distortion performance but, generally, by no more than 0.08 bit/pixel, compared with the much simpler Laplacian model-based quantizers and noiseless codes. This provides strong motivation for the selection of pyramid codes for transform and subband image coding.     

基于过采样抽取子带处理的宽带雷达杂波抑制

针对宽带雷达目标回波在相参处理时间内存在越距离单元走动的问题,提出了一种基于过采样抽取子带处理的宽带雷达杂波抑制与目标回波相参积累的方法。通过过采样离散傅立叶变换（DFT）调制滤波器组将宽带雷达回波分为若干子带,每个子带内目标回波不存在越距离单元走动,从而可以通过对相参回波序列的加权处理实现杂波抑制和目标回波的相参积累,将各子带的处理结果通过综合滤波器组重构得到相参积累后的目标距离像。由于采用了过采样抽取滤波器组,子带加权处理所产生的剩余混叠失真很小,因此,与传统的基于2维快速傅立叶变换（FFT）的处理方法相比,可以获得更大的信杂比改善因子和更小的重构距离像失真。通过理论分析和仿真实验证明了该方法对宽带雷达杂波抑制和目标回波相参积累的有效性。     

基于图像质量评价参数的FDST域图像融合

为了提升多源图像融合精度,提出了一种基于图像 质量评价参数的有限离散剪切波变换(FDST)域图像自适应融合方法。利用FDST对源图像进行 多尺度、多方向分解,低频子带图像采用结构相似度与空间频率两种图像评价参数作为系数 权值,高频子带图像应用区域空间频率取大的融合策略。应用有限离散剪切波逆变换(FDSIT )重 构图像。采用多组多源图像进行融合实验,并对融合结果进行了客观评价。实验结果表明, 本文提出的融合方法在主观和客观评价上均优于其他多尺度融合方法,具有更好的融合效果 。     

Interference cancellation in respiratory sounds via amultiresolution joint time-delay and signal-estimation scheme

This paper is concerned with the problem of cancellation of heart sounds from the acquired respiratory sounds using a new joint time-delay and signal-estimation (JTDSE) procedure. Multiresolution discrete wavelet transform (DWT) is first applied to decompose the signals into several subbands. To accurately separate the heart sounds from the acquired respiratory sounds, time-delay estimation (TDE) is performed iteratively in each subband using two adaptation mechanisms that minimize the sum of squared errors between these signals. The time delay is updated using a nonlinear adaptation, namely the Levenberg-Marquardt (LM) algorithm, while the function of the other adaptive system-which uses the block fast transversal filter (BFTF)-is to minimize the mean squared error between the outputs of the delay estimator and the adaptive filter. The proposed methodology possesses a number of key benefits such as the incorporation of multiple complementary information at different subbands, robustness in presence of noise, and accuracy in TDE. The scheme is applied to several cases of simulated and actual respiratory sounds under different conditions and the results are compared with those of the standard adaptive filtering. The results showed the promise of the scheme for the TDE and subsequent interference cancellation     

利用直接矩阵求逆和临界采样子带自适应滤波器的快速系统辨识

在许多应用中,子带自适应滤波器结构已经显示了其在计算和性能上的优点。基于最近提出的一个采用临界采样滤波器组的子带自适应结构,该文引入了子带直接矩阵求逆(DMI)算法。在保持了该算法快速收敛优点的同时,利用相关矩阵块三对角的特殊结构,降低了该算法的计算复杂度。理论分析及计算机实验显示,子带直接矩阵求逆算法只需经过较少的更新次数自适应子滤波器自由度的两倍,就能够收敛到高于最小均方误差的3dB附近。     

Mean-square performance analysis of the family of selective partial update and selective regressor affine projection algorithms

In this paper we present a general formalism for the establishment and mean-square performance analysis of the family of selective partial update affine projection (SPU-AP), selective regressor affine projection (SR-AP), and selective partial update subband adaptive filter (SPU-SAF) algorithms. This analysis is based on energy conservation arguments and does not need to assume a Gaussian or white distribution for the regressors. We demonstrate through simulations that the results are useful in predicting the performance of these adaptive filter algorithms.     

Best wavelet packet bases in a rate-distortion sense

A fast rate-distortion (R-D) optimal scheme for coding adaptive trees whose individual nodes spawn descendents forming a disjoint and complete basis cover for the space spanned by their parent nodes is presented. The scheme guarantees operation on the convex hull of the operational R-D curve and uses a fast dynamic programing pruning algorithm to markedly reduce computational complexity. Applications for this coding technique include R. Coefman et al.'s (Yale Univ., 1990) generalized multiresolution wavelet packet decomposition, iterative subband coders, and quadtree structures. Applications to image processing involving wavelet packets as well as discrete cosine transform (DCT) quadtrees are presented.     

Prioritized DCT for compression and progressive transmission ofimages

An approach is based on the block discrete cosine transform (DCT). The novelty of this approach is that the transform coefficients of all image blocks are coded and transmitted in absolute magnitude order. The resulting ordered-by-magnitude transmission is accomplished without sacrificing coding efficiency by using partition priority coding. Coding and transmission are adaptive to the characteristics of each individual image. and therefore, very efficient. Another advantage of this approach is its high progression effectiveness. Since the largest transform coefficients that capture the most important characteristics of images are coded and transmitted first, this method is well suited for progressive image transmission. Further compression of the image-data is achieved by multiple distribution entropy coding, a technique based on arithmetic coding. Experiments show that the approach compares favorably with previously reported DCT and subband image codecs.     

Synthesis filterbank with low hardware complexity for subband imagecoding

Different algorithms are evaluated and optimizations are performed to obtain a filterbank for subband coding of images especially suited for very large scale integration (VLSI) implementation. Based on a filterbank consisting of two finite impulse response (FIR) filters combined with an 8-point discrete cosine transform (DCT), we investigate how the quantization of filter coefficients and twiddle factors in different algorithms affects the quality of the filterbank. It is found that a DCT based on a Stasinksi (1989) algorithm with twiddle factors of only 5 b together with FIR filter coefficients of 10 b gives a filterbank with high coding gain, no blocking artifacts, and limited ringing. The VLSI complexity is comparable to that of DCT transforms.     

基于子带SDL的宽带自适应波束形成

针对宽带波束形成通常需要大量阵元或延迟线所带来的硬件开销较大,波束形成效率相对较低的问题,提出一种基于子带阵元延迟线（SDL）的宽带自适应波束形成算法。该算法首先建立子带SDL模型,然后利用分析滤波器组将阵元接收的宽带信号分解为子带信号并进行相应的子带线性约束最小方差（LCMV）波束形成,最后通过综合滤波器组得到全带的波束形成。仿真结果表明,子带波束形成不仅具有比全带波束形成更高的效率,更好的频率不变性,更强的抗干扰能力及更快的收敛速度,而且可以大大降低硬件开销。