On the computation of running discrete cosine and sine transform

Two algorithms are given for the computation of the updated discrete cosine transform-II (DCT-II), discrete sine transform-II (DST-II), discrete cosine transform-IV (DCT-IV), and discrete sine transform-IV (DST-IV). It is pointed out that the algorithm used for running DCT-IV can also be used for computation for running DST-IV without additional computational overhead. An architecture which is common and suitable for VLSI implementation of the derived algorithms is also presented. Preliminary studies have shown that the architecture can easily be implemented in VLSI form, and, in conjunction with a high-speed digital signal processor (for example ADSP 2100A), it can be used for real-time transform domain LMS adaptive filtering (128 taps) of 8 kHz sample rate speech signals     

基于变换域全相位FIR自适应滤波算法

基于一种全相位FIR自适应滤波器,将重叠滤波思想引入变换域LMS算法,提出了DFT、DCT和DST变换域的带窗重叠自适应滤波算法(WO-TLMS).与传统的变换域LMS(TLMS)算法相比,WO-TLMS算法提高了收敛速度同时具有较低的稳态均方误差.理论分析了算法的收敛性,实验中通过和TLMS算法的比较验证了WO-TLMS算法的优越性.     

离散小波变换域LMS自适应消噪

本文按照自成体系的离散小波变换正交分解框架,直接从离散域出发,得出离散小波变换域ＬＭＳ（ＤＷＬＭＳ）,变换的正交性与采样率无关,并可使用Ｍａｌｌａｔ快速算法。方法可减小自适应应滤波器输入向量自相关阵的谱动态范围,提出了传统ＬＭＳ算法的收敛速度和稳定性。     

LFM信号的分数阶傅里叶域自适应滤波算法研究

对于线性调频信号(LFM)的滤波,采用处理平稳信号的方法对其滤波往往得不到很好的效果。本文利用了线性调频信号在分数傅里叶变换域上具有很好的时频聚焦性的特点,来实现信号在分数阶傅里叶域的自适应滤波,自适应滤波算法采用改进的步长LMS方法,对传统的LMS算法做出了改进,算法中步长处理中引入了一个限制因子,可以较好地解决算法收敛速度和稳态失调量之间的矛盾。仿真结果表明,此算法在处理分数阶域的LFM信号滤波比传统的LMS算法有较好的滤波效果。      

一种新的变步长变换域自适应滤波算法

变换域LMS算法能通过正交变换有效降低输入信号自相关矩阵特征值的分散程度,可提高算法的收敛速度;变步长LMS算法可以克服固定步长因子所导致的算法在较快收敛速度和较小稳态误差之间存在的矛盾,从而获得较快的收敛速度和较好的收敛结果。将二者相结合,提出了一种新的变步长变换域自适应滤波算法。计算机仿真结果表明该算法具有更快的收敛速度和更小的稳态误差,并且运算量较少,具有良好的实用性能。     

一种基于变换域LMS的多用户检测算法

LMS算法收敛速度由输入自相关矩阵特征值的发散程度决定,很多场合下收敛速度慢不能满足需求。变换域自正交LMS算法在变换域对输入信号做正交处理,使得输入自相关矩阵变成单位矩阵,能够有效地提高收敛速度。研究了基于变换域LMS的多用户检测算法,仿真表明该算法能够提高多用户检测器的性能。     

基于小波变换变步长LMS自适应信道均衡的新方法

结合变换域最小均方(LMS)和变步长LMS算法的优势,提出了一种基于小波变换的变步长LMS自适应均衡方法。该方法中步长调整函数采用了改进的Sigmoid函数,该函数具有简单且误差信号接近零时变化缓慢的特点。并且,在训练模式、判决引导模式以及混合模式下,将提出方法和传统均衡方法进行了仿真比较。结果表明,所提出的方法比传统的线性LMS算法、变步长LMS以及小波变换LMS收敛更快、性能更优。     

基于自适应分数阶傅里叶变换的线性调频信号检测及参数估计

该文提出了一种基于最小均方算法的自适应计算分数阶傅里叶变换的方法并将该方法应用到多分量chirp信号的检测与估计之中。该方法通过对连续型分数阶傅里叶反变换进行离散化采样,得到适合数值计算的离散形式,进而通过适当的选择输入向量和目标函数构造自适应滤波器,经过最小均方算法进行训练后所得的滤波器权系数即为分数阶傅里叶变换的结果。仿真实验表明,该方法可以用来计算分数阶傅里叶变换及对chirp信号进行检测和参数估计,且计算延时相对较小。     

Combined RLS-LMS initialization for per tone equalizers in DMT-receivers

In discrete multitone receivers, the classical equalizer structure consists of a (real) time domain equalizer (TEQ) combined with complex one-tap frequency domain equalizers. An alternative receiver is based on a per tone equalization (PTEQ), which optimizes the signal-to-noise ratio (SNR) on each tone separately and, hence, the total bitrate. In this paper, a new initialization scheme for the PTEQ is introduced, based on a combination of least mean squares (LMS) and recursive least squares (RLS) adaptive filtering. It is shown that the proposed method has only slightly slower convergence than full square-root RLS (SR-RLS) while complexity as well as memory cost are reduced considerably. Hence, in terms of complexity and convergence speed, the proposed algorithm is in between LMS and RLS.     

基于小波变换的变步长LMS自适应滤波算法

变步长LMS自适应滤波算法通过构造合适的步长因子有效的解决了传统LMS算法收敛速度和稳态误差相矛盾的问题.变换域LMS自适应滤波算法通过正交变换降低了输入信号矩阵的相关性,提高了算法的收敛速度.将这两种算法相结合,提出了一种新的基于小波变换的变步长LMS自适应滤波算法.仿真结果表明,该算法无论是收敛速度还是稳态误差都有了很大的提高.     

一种基于多尺度小波变换的自适应滤波新算法

将多尺度小波变换的理论引入到LMS自适应滤波器的设计中,分析了基于多尺度正交小波变换的自适应滤波算法的原理;将变步长LMS算法与多尺度小波变换的思想结合,提出了一种新的小波自适应滤波算法(MSWT-MVSS-LMS),新算法既减少了输入向量自相关矩阵条件数,又克服了固定步长LMS算法在收敛速度与收敛精度方面与步长因子μ的矛盾,获得了更好的收敛速度和稳定性.仿真结果表明新算法是有效的和优越的.     

一种新的变换域自适应LMS滤波算法

为了改善时变系统中的LMS算法收敛速度,一般可以在变换域进行自适应处理。通过研究和分析分数阶傅里叶变换与时-频平面的关系,提出在分数阶傅里叶变换域进行自适应时-频滤波。所提出的方法首先搜索最佳变换域,然后在分数阶傅里叶变换域进行LMS自适应滤波。仿真结果表明,与目前一些基于变换域的方法对比,新方法通过对时-频平面的旋转,可以显著加速算法收敛性。     

基于LMS算法的多点滑动DFT方法

 根据自适应LMS法,LMS谱分析器可以通过递归运算完成滑动窗口中数据的DFT运算.本文推导了LMS算法及多点滑动DFT运算之间的关系式,并由此提出了一种基于LMS算法的多点滑动DFT运算方法.文章在理论推导的同时,进行了计算机仿真验证.该方法使用方便,可灵活适用于不同的滑动窗口大小及滑动步长参数.     

Generalized sliding FFT and its application to implementation ofblock LMS adaptive filters

This paper has two contributions. First, the concept of the generalized sliding fast Fourier transform (GSFFT) as an efficient implementation of the hopping FFT is introduced. Application of the GSFFT is broad and not limited to what has been considered in this paper. The frequency domain block LMS (FBLMS) adaptive filters are then revised, and their implementations for block lengths less than the length of the adaptive filter are studied. The GSFFT and the available pruned FFTs are used to give an efficient implementation of these filters. In the particular case of the block length equal to one, where the FBLMS algorithm reduces to the frequency domain LMS (FLMS) algorithm, it is shown that the latter can be implemented with the order of M complexity, where M is the length of the adaptive filter     

LMS算法的二次稳定性及鲁棒LMS算法

本文在时域内研究LMS算法(least mean square algorithm)的稳定性及鲁棒LMS算法的构造.首先将LMS算法表达式转化为标准的离散时间系统状态方程形式,之后运用线性矩阵不等式(LMI)技术对其二次稳定性进行了分析.针对滤波过程中会出现的输入和测量噪声干扰,本文提出了一种兼顾收敛性、鲁棒稳定性以及鲁棒性能的鲁棒LMS算法,最后给出了仿真算例,通过和一般的LMS算法的比较,体现了这种鲁棒LMS算法的优越性.     

Channel estimation techniques based on pilot arrangement in OFDM systems

Channel estimation techniques for OFDM systems based on a pilot arrangement are investigated. Channel estimation based on a comb type pilot arrangement is studied through different algorithms for both estimating the channel at pilot frequencies and interpolating the channel. Channel estimation at pilot frequencies is based on LS and LMS methods while channel interpolation is done using linear interpolation, second order interpolation, low-pass interpolation, spline cubic interpolation, and time domain interpolation. Time-domain interpolation is obtained by passing to the time domain by means of IDFT (inverse discrete Fourier transform), zero padding and going back to the frequency domain by DFT (discrete Fourier transform). In addition, channel estimation based on a block type pilot arrangement is performed by sending pilots in every sub-channel and using this estimation for a specific number of following symbols. We have also implemented a decision feedback equalizer for all sub-channels followed by periodic block-type pilots. We have compared the performances of all schemes by measuring bit error rates with 16QAM, QPSK, DQPSK and BPSK as modulation schemes, and multipath Rayleigh fading and AR based fading channels as channel models.     

Application of running Hartley transform in adaptive digitalfiltering

A transform-domain adaptive digital filtering technique based on the concept of the running discrete Hartley transform (RDHT) is presented, with an application. This Hartley implementation is shown to have a speed advantage over the running FFT approach proposed and implemented recently. Possible implementation of the RDHT in various applications is outlined. Particular attention is given to implementation of the RDHT in the adaptive nonrecursive LMS algorithm     

A class of frequency-domain adaptive approaches to blind multichannel identification

We extend our previous studies on adaptive blind channel identification from the time domain into the frequency domain. A class of frequency-domain adaptive approaches, including the multichannel frequency-domain LMS (MCFLMS) and constrained/unconstrained normalized multichannel frequency-domain LMS (NMCFLMS) algorithms, are proposed. By utilizing the fast Fourier transform (FFT) and overlap-save techniques, the convolution and correlation operations that are computationally intensive when performed by the time-domain multichannel LMS (MCLMS) or multichannel Newton (MCN) methods are efficiently implemented in the frequency domain, and the MCFLMS is rigorously derived. In order to achieve independent and uniform convergence for each filter coefficient and, therefore, accelerate the overall convergence, the coefficient updates are properly normalized at each iteration, and the NMCFLMS algorithms are developed. Simulations show that the frequency-domain adaptive approaches perform as well as or better than their time-domain counterparts and the cross-relation (CR) batch method in most practical cases. It is remarkable that for a three-channel acoustic system with long impulse responses (256 taps in each channel) excited by a male speech signal, only the proposed NMCFLMS algorithm succeeds in determining a reasonably accurate channel estimate, which is good enough for applications such as time delay estimation.     

Block-based SVD image watermarking in spatial and transform domains

The idea of this paper is to implement an efficient block-by-block singular value (SV) decomposition digital image watermarking algorithm, which is implemented in both the spatial and transforms domains. The discrete wavelet transform (DWT), the discrete cosine transform and the discrete Fourier transform are exploited for this purpose. The original image or one of its transforms is segmented into non-overlapping blocks, and consequently the image to be inserted as a watermark is embedded in the SVs of these blocks. Embedding the watermark on a block-by-block manner ensures security and robustness to attacks such like Gaussian noise, cropping and compression. The proposed algorithm can also be used for colour image watermarking. A comparison study between the proposed block-based watermarking algorithm and the method of Liu is performed for watermarking in all domains. Simulation results ensure that the proposed algorithm is more effective than the traditional method of Liu, especially when the watermarking is performed in the DWT domain.     

DSSS通信中基于快速更新子带自适应滤波的窄带干扰抑制

本文面向直接序列扩频(DSSS)通信中的窄带干扰抑制,将分块更新子带自适应滤波的高频谱分隔特性和直接变换自适应滤波的逐点更新特性结合起来,提出了一种快速更新子带自适应(FRSAF)算法,给出了算法的迭代因子收敛界和快速实现结构。理论分析表明：该算法收敛迅速、迭代稳健,其性能明显优于经典子带自适应滤波算法和DCT/DFT-LMS算法,应用于DSSS通信可以得到优良的干扰抑制效果。仿真结果验证了上述结论。