Recursive time-varying filter banks for subband image coding

Filter banks, subband/wavelets, and multiresolution decompositions that employ recursive filters have been considered previously and are recognized for their efficiency in partitioning the frequency spectrum. This paper presents an analysis of a new infinite impulse response (IIR) filter bank in which these computationally efficient filters may be changed adaptively in response to the input. The new filter bank framework is presented and discussed in the context of subband image coding. In the absence of quantization errors, exact reconstruction can be achieved. By the proper choice of an adaptation scheme, it is shown that recursive linear time-varying (LTV) filter banks can yield improvement over conventional ones.     

Synthesis filter bank optimization in two-dimensional separablesubband coding systems

Subband coding is a popular and well established technique used in visual communications, such as image and video transmission. In the absence of quantization and transmission errors, the analysis and synthesis filters in a subband coding scheme can be designed to obtain perfect reconstruction of the input signal, but this is no longer the optimal solution in the presence of quantization of the subband coefficients. We presuppose the use of a two-dimensional (2-D) separable subband scheme and we address the problem of designing, for a given analysis filter bank and assuming uniform quantization of the subband coefficients, the set of row and column synthesis filters that minimize the mean squared reconstruction error at the output of the subband system. Since the corresponding optimization problem is inherently nonlinear, we propose a suboptimal solution that extends a one-dimensional (l-D) optimal filter design procedure, already presented in the literature, to a 2-D separable synthesis filter bank. The separable 2-D extension is not trivial, since the processing in one direction, e.g., the rows, alters the statistics of the signals for the design of the filters in the other direction, e.g., the columns. To further simplify the filter design, we propose to model the input image as a 2-D separable Markov process plus an additive white component. Several design examples using both synthetic signals and real world images are presented, showing that the filters designed using the proposed technique can give a significant gain with respect to the perfect reconstruction solution, especially when the dither technique is used for quantization. The simulation results also show that the proposed image model can be conveniently used in the synthesis filter design procedure.     

Design of hybrid filter banks for analog/digital conversion

This paper presents design algorithms for hybrid filter banks (HFBs) for high-speed, high-resolution conversion between analog and digital signals. The HFB is an unconventional class of filter bank that employs both analog and digital filters. When used in conjunction with an array of slower speed converters, the HFB improves the speed and resolution of the conversion compared with the standard time-interleaved array conversion technique. The analog and digital filters in the HFB must be designed so that they adequately isolate the channels and do not introduce reconstruction errors that limit the resolution of the system. To design continuous-time analog filters for HFBs, a discrete-time-to-continuous-time (“Z-to-S”) transform is developed to convert a perfect reconstruction (PR) discrete-time filter bank into a near-PR HFB; a computationally efficient algorithm based on the fast Fourier transform (FFT) is developed to design the digital filters for HFBs. A two-channel HFB is designed with sixth-order continuous-time analog filters and length 64 FIR digital filters that yield -86 dB average aliasing error. To design discrete-time analog filters (e.g., switched-capacitors or charge-coupled devices) for HFBs, a lossless factorization of a PR discrete-time filter bank is used so that the reconstruction error is not affected by filter coefficient quantization. A gain normalization technique is developed to maximize the dynamic range in the finite-precision implementation. A four-channel HFB is designed with 9-bit (integer) filter coefficients. With internal precision limited to the equivalent of 15 bits, the maximum aliasing error is -70 dB, and with the equivalent of 20 bits internal precision, maximum aliasing is -100 dB. The 9-bit filter coefficients degrade the stopband attenuation (compared with unquantized coefficients) by less than 3 dB     

A multidimensional wave digital filter bank for video-based motion analysis

Multidimensional wave digital filters (MDWDF) exhibit the same desirable properties as 1D WDFs, most notably including passivity and therefore guaranteed stability as well as high robustness. A possible application for such MDWDFs may be found in motion analysis of image sequences by means of filters with fan-shaped transfer functions, where content with specific movement information can be extracted. For that matter, a parallel filter bank is needed to differentiate object motion into separate classes. In this paper, a new specialized MDWDF fan filter structure is introduced, possessing both reduced computational complexity and memory requirements compared to existing approaches. Additionally, part of the processing can be shared among all bands, further increasing efficiency.     

调制混合滤波器组的宽带模拟信号采样研究

对于超宽带模拟信号,很难用单个模拟数字转换器（ADC）直接进行采样。该文提出了一种新的并行调制混合滤波器组结构用于实现超宽带模拟信号的采样,首先,将每一路宽带模拟输入信号进行余弦调制,并用相同的低通模拟滤波器均匀分割输入信号的带宽;然后,采用相同的ADC将子带信号数字化;各路子带信号通过上采样器后用数字综合滤波器综合得到原宽带模拟输入信号的数字重构。综合滤波器采用总体最小二乘准则下的特征值滤波器设计方法得到。本文所提出的系统结构不需要使用高速的采样保持电路,降低了系统实现的难度,并且设计的系统具有与其它混合滤波器组相近的重构性能。仿真结果表明了本方法的有效性。      

基于代价参考粒子滤波器组的检测前跟踪算法

基于粒子滤波的检测前跟踪方法是检测和估计非线性调频信号的有效方法之一。但此类方法运算量大,难以并行执行。此外,由于粒子滤波算法收敛较慢,基于粒子滤波的检测前跟踪方法的检测和状态估计能力有待提高。针对上述问题,该文首先提出一种代价参考粒子滤波器组。该滤波器组收敛快速,具有完全的并行结构,可快速准确地估计非线性调频信号的瞬时频率。其次,提出基于代价参考滤波器组的检测前跟踪算法,可在给定虚警率下,在各个时刻检测目标和估计目标状态。两类非线性调频信号检测和估计的仿真结果表明,基于代价参考粒子滤波器组的检测前跟踪算法的检测性能、估计性能和运行速率均优于类似的方法,如基于粒子滤波的检测前跟踪方法,基于Rutten粒子滤波的检测前跟踪方法等。     

二维频域内的运动目标检测方法

传统频域内的运动目标检测算法通常需要设计三维时空滤波器组来实现,存在待选参数多、滤波器设计难度大等问题。文中提出了一种二维频域内的运动目标检测方法,通过对动态图像的行列分解,将三维频域内的运动检测问题转化到两组二维频域内进行,从而降低了滤波器设计的难度。给出了一种提取主运动能量的自适应滤波算法,通过剔除背景和噪声的频率成分,有效地检测出运动目标。仿真结果表明,该方法可有效降低背景配准误差和噪声的影响。     

临界采样余弦调制滤波器组算法综述

在多速率数字信号处理系统中,余弦调制滤波器组受到广泛关注.它是一种常用的滤波器组,可以通过对低通原型滤波器优化并进行余弦调制得到分析和综合滤波器组,并且可以做到对信号的完全重构,具有计算复杂度低和设计简单等优点.该文综述了余弦调制滤波器组的发展过程及各种算法.     

Two-channel IIR QMF banks with approximately linear-phase analysisand synthesis filters

Perfect linear-phase two-channel QMF banks require the use of finite impulse response (FIR) analysis and synthesis filters. Although they are less expensive and yield superior stopband characteristics, perfect linear phase cannot be achieved with stable infinite impulse response (IIR) filters. Thus, IIR designs usually incorporate a postprocessing equalizer that is optimized to reduce the phase distortion of the entire filter bank. However, the analysis and synthesis filters of such an IIR filter bank are not linear phase. In this paper, a computationally simple method to obtain IIR analysis and synthesis filters that possess negligible phase distortion is presented. The method is based on first applying the balanced reduction procedure to obtain nearly allpass IIR polyphase components and then approximating these with perfect allpass IIR polyphase components. The resulting IIR designs already have only negligible phase distortion. However, if required, further improvement may be achieved through optimization of the filter parameters. For this purpose, a suitable objective function is presented. Bounds for the magnitude and phase errors of the designs are also derived. Design examples indicate that the derived IIR filter banks are more efficient in terms of computational complexity than the FIR prototypes and perfect reconstruction FIR filter banks. Although the PR FIR filter banks when implemented with the one-multiplier lattice structure and IIR filter banks are comparable in terms of computational complexity, the former is very sensitive to coefficient quantization effects     

Blind multipath separation and combining technique for signal recovery

To achieve better mitigation of both cochannel interference (CCI) and intersymbol interference, a new structure using generalized estimation of multipath signals in conjunction with maximal-ratio combining diversity for wireless communications over multipath channels is introduced. In this structure, the signal replicas received from multiple paths are first independently produced by a bank of blind spatial filters and then constructively combined by a diversity combining receiver for final signal estimate. The new scheme can be applied on single antenna array or between multiple antenna subarrays. It will be shown, from both theoretical analysis and numerical experiments, that the new scheme provides both space diversity gains and path diversity gains while suppressing the CCIs.     

Mean-square performance of a convex combination of two adaptive filters

Combination approaches provide an interesting way to improve adaptive filter performance. In this paper, we study the mean-square performance of a convex combination of two transversal filters. The individual filters are independently adapted using their own error signals, while the combination is adapted by means of a stochastic gradient algorithm in order to minimize the error of the overall structure. General expressions are derived that show that the method is universal with respect to the component filters, i.e., in steady-state, it performs at least as well as the best component filter. Furthermore, when the correlation between the a priori errors of the components is low enough, their combination is able to outperform both of them. Using energy conservation relations, we specialize the results to a combination of least mean-square filters operating both in stationary and in nonstationary scenarios. We also show how the universality of the scheme can be exploited to design filters with improved tracking performance.     

Analysis of compressive receivers for the optimal interception offrequency-hopped waveforms

Establishes that the compressive receiver is a practical interceptor of high performance. Given a signal of a particular duration, a compressive receiver can estimate simultaneously all frequency components within a set wide band. This processing is similar to a parallel bank of narrowband filters, which is the optimal detector of frequency-hopped signals. Furthermore, hop frequency is estimated to yield performance equal to the parallel filter configuration. The authors assume interference to be stationary, colored Gaussian noise, and present a model of the compressive receiver that contains all its salient features. Low energy coherence detection is achieved by taking the compressive receiver output as an observation and applying likelihood ratio theory at small signal-to-noise ratios. For small signals, this approach guarantees the largest probability of correct detection for a given probability of false alarm, and thus provides a reference, to which simplified or ad hoc schemes can be compared. Since the low energy coherence detector has an unwieldy structure, a simplified suboptimal detector structure is developed that consists of a simple filter, followed by a sampler and a square-envelope detector. Several candidates for the filter's response are presented. The performance of the low energy coherence detector based on compressive receiver observations is compared to the optimal filter-bank detector based on direct observations, thus showing the exact loss incurred when a compressive receiver is used. The performance of various simplified schemes, based on compressive receiver observations, is analyzed     

An integrated framework for adaptive subband image coding

Previous work on filter banks and related expansions has revealed an interesting insight: different filter bank trees can be regarded as different ways of constructing orthonormal bases for linear signal expansion. In particular, fast algorithms for finding best bases in an operational rate-distortion (R/D) sense have been successfully used in image coding. Independently of this work, other research has also explored the design of filter banks that optimize energy compaction for a single signal or a class of signals. In this paper, we integrate these two different but complementary approaches to best-basis design and propose a coding paradigm in which subband filters, tree structure, and quantizers are chosen to optimize the R/D performance. These coder attributes represent side information. They are selected from a codebook designed off-line from training data, using R/D as the design criterion. This approach provides a rational framework in which to explore alternatives to empirical design of filter banks, quantizers, and other coding parameters. The on-line coding algorithm is a relatively simple extension of current R/D-optimal coding algorithms that operate with fixed filter banks and empirically designed quantizer codebooks. In particular, it is shown that selection of the best adapted filter bank from the codebook is computationally elementary     

Ordered Successive Interference Cancellation (OSIC) in V-BLAST Systems with Asynchronous Transmission Mode

In this paper, we first present a successive interference cancellation (SIC) architecture for the V-BLAST system with asynchronous transmission mode, which is proposed in Shao and Tang (IEEE Trans Veh Technol 56(6):3827–3837, 2007) and characterized by achieving maximum possible diversity only by using a simple linear detection scheme under zero forcing (ZF) criterion. The SIC architecture that intuitively derived form the mathematical expression of the sampled output vector of a bank of matched filters at each receive antenna has a very low complexity, however, it can not employ the optimal detection order to minimize the propagation of errors from one step of detection to the next. Furthermore we find that it actually consumes extra energy of the previously detected signals to recover a signal. Therefore a ordered successive interference cancellation (OSIC) architecture is subsequently proposed to overcome the two shortcomings mentioned above. Simulations demonstrate that the OSIC in the asynchronous V-BLAST system can give much better performance at a little cost of complexity than it dose in conventional synchronous scheme.     

Vector space framework for unification of one- and multidimensionalfilter bank theory

A number of results in filter bank theory can be viewed using vector space notations. This simplifies the proofs of many important results. In this paper, we first introduce the framework of vector space, and then use this framework to derive some known and some new filter bank results as well. For example, the relation among the Hermitian image property, orthonormality, and the perfect reconstruction (PR) property is well-known for the case of one-dimensional (1-D) analysis/synthesis filter banks. We can prove the same result in a more general vector space setting. This vector space framework has the advantage that even the most general filter banks, namely, multidimensional nonuniform filter banks with rational decimation matrices, become a special case. Many results in 1-D filter bank theory are hence extended to the multidimensional case, with some algebraic manipulations of integer matrices. Some examples are: the equivalence of biorthonormality and the PR property, the interchangeability of analysis and synthesis filters, the connection between analysis/synthesis filter banks and synthesis/analysis transmultiplexers, etc. Furthermore, we obtain the subband convolution scheme by starting from the generalized Parseval's relation in vector space. Several theoretical results of wavelet transform can also be derived using this framework. In particular, we derive the wavelet convolution theorem     

The optimal linear receiving filter for digital transmission overnonlinear channels

The filter minimizes the mean-square error between the sampled input to the decision device and the corresponding transmitted symbols. It is shown that the structure of this optimal receiving filter corresponds to a bank of matched filters, each followed by a linear transversal filter. Applications of this theory cover nonlinear channels with memory, linearly or nonlinearly modulated signals, and disturbances other than Gaussian noise (e.g. interference from other data sources)     

Method of Forming Classified Training Sample in Case of Spacial Signal Processing under Influence of Combined Interference

Under the conditions of the combined interference, the operation efficiency of the radar equipment is substantially deteriorated. This is due to the decorrelation of the signals of the point source of active interference acting on the radar by passive interference. In this article methods of formation of the classified training sample, generated only by active interference, are considered for adaptation of the weight coefficients of spatial filters under conditions of combined interference presence. An effective method of forming the classified training sample generated by active masking interference was developed for spatial processing of radar signals under conditions of simultaneous exposure to passive interference. The developed method of forming the training sample is based on estimating the width of the normalized autocorrelation function in each element of the distance resolution. The current analysis of the combined interference components in each resolution element improves the quality of the interference component classification and, as a result, minimizes the effect of passive interference on the adaptation process of the spatial filter. The theoretical and practical aspects of the formation of the classified training sample are considered. The functional scheme of the classifier for the combined interference components is developed. The efficiency of the proposed method is compared with known correlation methods. The current analysis of the combined interference components in each element of the range resolution improves the quality of interference classification, which is important in the context of complex hydrometeorological conditions.     

New Low Bit-Rate Audio Coding Approach Based on Adaptive Wavelet Analysis

This paper describes a new audio coding scheme based on adaptive wavelet analysis that provides transparent audio coding for CD-audio signals at low bit rates (≈1.4 bits/sample per channel). A new perceptual cost function is defined to obtain the best wavelet-packet base for each audio frame. The sharp variations in quantization noise that appear at the border of the frames are minimized by a novel approach that avoids overlapping. The proposed coder guarantees high perceptual quality using filters that generate wavelets of any compact support, because a bit-allocation algorithm that takes into account the equivalent filter frequency responses of the synthesis filter bank branches is used.     

D/A转换在System View环境下的仿真及硬件实现

针对模拟信号在传输介质中优于数字信号,而设计数模转换模块。首先用System View对DAC模块进了仿真。然后设计的D/A转换的硬件电路。通过设计了一个前置的串并转换电路,不仅可以实现8位并行数字信号的D/A转换,还可实现8位串口输入数字信号的D/A转换。在输出端,接入一个有源二阶低通滤波电路,使模拟输出更为平滑。达到了在实际范围内较低波形衰减的目的。