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Description: Title: MMSE Receiver for DS-SS in AWGN Channel
Author: Panson Tantikovit
Summary: An adaptive receiver for DS-SS systems
MATLAB Release: R12.1
Required Products: Communications Toolbox,Signal Processing Blockset
Description: This is an adaptive receiver for a direct-sequence spread spectrum (DS-SS) system over an AWGN channel. The adaptive receiver block is modified from the LMS adaptive filter block in DSP Blockset. For DS-SS signal reception, the adaptive filter needs to have multi-rate operation. The input sample rate is equal to chip rate and the output is at symbol rate. Two rates are related by PG, processing gain.
-Title: MMSE Receiver for DS-SS in AWGN Channel Author: Panson Tantikovit Summary: An adaptive receiver for DS-SS systems MATLAB Release: R12.1 Required Products: Communications Toolbox,Signal Processing Blockset Description: This is an adaptive receiver for a direct-sequence spread spectrum (DS-SS) system over an AWGN channel. The adaptive receiver block is modified from the LMS adaptive filter block in DSP Blockset. For DS-SS signal reception, the adaptive filter needs to have multi-rate operation. The input sample rate is equal to chip rate and the output is at symbol rate. Two rates are related by PG, processing gain.
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Author: zzp |
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Description: 用于语音回波消除的lms算法的改进，使用这种方法后，能够有效的加快回波消除功能，并且效果也不错，已经在dsp程序里实现-voice echo canceller for the LMS algorithm, the use of such methods, to accelerate the effective echo canceller function, and also worked well, in achieving procedures dsp
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Author: 吴玉山 |
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Description: DSP II作业，有关LMS的计算机练习题-II DSP operations, the LMS Computer Exercises
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Size: 2562 |
Author: 周姗姗 |
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Description: 论文主要研究了自适应滤波器的基本结构和最小均方误差算法(LMS算法)，同时采用MATLAB语言对该算法进行了仿真，然后使用SY-5402EVM 评估板对其进行了DSP 实现，并对结果进行了分析。
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Size: 501354 |
Author: 海德 |
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Description: 主要 研 究 用于声音通讯系统中的声回波抵消技术。声回波抵消通常采用声回波抵
消器来实现。具体方法是用自适应滤波器来估计回波信号，并从麦克风信号中减掉该
估计值，从而实现声回波的抵消。回波抵消器中常用的几种自适应滤波算法有LMS,
NLMS, RLS等算法。对现有主要算法的性能进行了分析，并对优缺点进行评价和比
较。为了在收敛速度和运算量之间得到很好的折衷，对NLMS算法改进，得到了
PNLMS算法，对传统的快处理块算法改进，得到了精确块算法。综合两个算法，获
得了一种基于回波路径冲击响应稀疏性和精确块技术的自适应算法(BEPNLMS),
并对该算法作了详细的分析和研究。针对具体应用以及DSP芯片的特点，结合
BEPNLMS算法，给出了一种基于TMS320VC5402 DSP实现回波抵消的软硬件设计
方案。对主要模块电路的设训作了详细介绍，画出了自适应算法和语音检测流程图，
写出了各模块程序的主要部分。最后，对新算法的收敛性和计算复杂度等主要性能作
了验证，说明本算法可行并满足各项指标要求。
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Author: hbeishang |
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Description: matlab在信号处理中的应用,主要有LMS，KERMAN滤波等常用算法-THIS IS A APLICATION OF DSP DEVELOPED BY MATLAB
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Author: 110500617 |
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Description: 自适应权向量法(AWVM) 检测弱信号常用LMS滤波算法. 这种方法的最大缺陷之
一是滤波器的权系数仅根据总误差值来调整的. 因此, 这些权系数相互有牵连, 这是LMS自
适应滤波器收敛响应较慢的原因之一. 克服这一缺点的一种方法是采用格型滤波算法. 应用
格型LSL滤波于AWVM中检测噪声中的高频弱信号, 并应用DSP实现高速数据处理, 仿真结
果证明了它的可行性和实时性及优越性.-Adaptiveweightvectormethod (AWVM) indetectingweaksignaloftenusesLMSalgo 2
rithm1Theseverelimitationofthealgorithmisthatweightcoefficientsofthefilterareadjustedonlyus 2
ingtotalerrorvalue 1Forthisreason,thoseweightcoefficientsarerelatedwitheachother,whichis
oneofthereasonsoftheslowconvergenceresponseofLMSalgorithm.Onemethodovercomingthe
faultistouselatticealgorithm.ThispaperapplieslatticeLSLalgorithmintoAWVMfordetectinghigh
frequencyweaksignalfromnoiseandforrealizinghigh 2speeddataprocessing 1Thesimulatedresults
showitsfeasibilityandrealityaswellassuperiority
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Author: 李丽 |
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Description: Readme
Complex class is very similar to the Standard C++ library complex class and the standard C++ class could be substituted in many cases.
Code is wrapped in namespace SPUC
SPUC DSP C++ source code library directory structure
array - Array class from TNT library
comm - primarily digital communications related classes
equalizers - Maximum Likelihood Sequence Detector, RLS and LMS algorithms
generic - most of the code/algorithms
functions - mostly functions (as opposed to classes)
matrix - Matrix/Vector class from IT++
specializations - mostly variations of basic blocks in generic
utility - classes to aid in debug
examples - Examples directory including
qpsk - QPSK + QPSK_VR examples-Readme
Complex class is very similar to the Standard C++ library complex class and the standard C++ class could be substituted in many cases.
Code is wrapped in namespace SPUC
SPUC DSP C++ source code library directory structure
array - Array class from TNT library
comm - primarily digital communications related classes
equalizers - Maximum Likelihood Sequence Detector, RLS and LMS algorithms
generic - most of the code/algorithms
functions - mostly functions (as opposed to classes)
matrix - Matrix/Vector class from IT++
specializations - mostly variations of basic blocks in generic
utility - classes to aid in debug
examples - Examples directory including
qpsk - QPSK + QPSK_VR examples
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Size: 429056 |
Author: Prabhat |
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Description: 讲述一种基于DSP的LMS基本原理，以及基本的参考资料-sedBasic principle about a DSP-ba LMS, as well as the basic reference
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Author: hubang |
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Description: 用于DSP的LMS算法源代码。采用C语言编写-For the DSP the LMS algorithm source code. Using C language
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Size: 1024 |
Author: 张辉 |
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Description: lms algorithm in dsp matlab
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Size: 1024 |
Author: ram |
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Description: LMS自适应算法，权的迭代及画性能表面的等高线绘图。-LMS，dsp
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Author: 凈嚨唲 |
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Description: LMS DSP的实现 CCS的工程 单点抽样 可用于滤波 预测等-LMS DSP implementation of CCS projects can be used for single-point sampling filtering prediction
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Size: 57344 |
Author: 孙建兴 |
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Description: 任务：
1） 借助MATLAB画出误差性能曲面和误差性能曲面的等值曲线（参考PPT2.1第17页的两幅图）；
2） 写出最陡下降法以及LMS算法的计算公式（取 ）；
3） 用MATLAB产生方差为0.05, 均值为0白噪音S(n)，并画出某次采样得到的波形（即产生任意一个噪声随机序列）；
4） 根据 2）中的公式，并利用 3）中产生的S(n)，在 1）中的误差性能曲面的等值曲线上叠加画出采用最陡下降法以及LMS法时H(n)的在叠代过程中的轨迹曲线（参考PPT2.1第17页的右下图的曲线1和曲线2）。
5）用MATLAB计算并画出LMS法时 随时间n的变化曲线（对应S(n)的某一次的一次实现）和e(n)波形；
注意：某一次实现的结果并不能从统计的角度反映实验的结果 的正确性，为得到具有统计特性的实验结果，可用足够多次的 实验结果的平均值作为实验的结果。用MATLAB计算并画出 采用LMS法时，J(n)的100次实验结果的平均值随时间n的变 化曲线（即 生成随机噪声信号并计算结果，重复执行100次， 求平均结果）。
6）在 1）中的误差性能曲面的等值曲线上，叠加画出采用LMS法得到的100次实验中的H(n)的平均值的轨迹曲线；
-Task:
1 ) Draw with MATLAB error performance surface and surface contour error performance curve ( refer PPT2.1 two chart on page 17 )
2 ) Write the steepest descent method and the LMS algorithm formula ( take )
3 ) Using MATLAB to generate variance of 0.05 with a mean of 0 white noise S (n), and draw a particular sampling waveform ( ie, any noise generated random sequence )
4 ) based on 2 ) of the formula , and use 3 ) generated in S (n), in a ) the error performance curves superimposed on the surface contour plot using the steepest descent method and the LMS method when H (n) the iterative process in the trajectory curve ( refer PPT2.1 page 17 bottom right of the curves 1 and 2 ) .
5 ) using MATLAB LMS method is used calculate and draw the curve with time n (corresponding to S (n) of a first implementation of a ) and e (n) waveform
Note : in a first implementation of the results are not reflected from a statistical point of view of accuracy of the experimental results ,
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Author: 刘小六 |
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Description: DSP 戴礼荣的第一次DSP作业，借助M AT L A B 画出误差性能曲面和误差性能曲面的等值曲线；用M AT L A B 产生方差为0.05，均值为0 白噪音S(n)，并画出其中一次实现的
波形图；在误差性能曲面的等
值曲线上叠加画出采用最陡下降法， LMS 法时H(n)的在叠代过程中的轨迹曲线-Homework of Dai Lirong
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Author: 陈春红 |
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Description: 基于DSP的有缘噪声控制中的自适应算法 LMS算法-DSP-based affinity for noise control in the adaptive algorithm LMS algorithm
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Author: 刘永平 |
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Description: 归一化LMS自适应滤波算法的matlab仿真及DSP实现-Normalized LMS adaptive filter algorithm MATLAB simulation and DSP Implementation
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Author: 唐鹏 |
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Description: 使用LMS方法对DSP数据进行去噪音处理，比较理想-Audio Denoising by lms method
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Size: 1024 |
Author: wangzy |
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Description: C语言自适应滤波算法，可移植到任意单片机上或其他工程项目代码上。在DSP28335通过验证。(The C adaptive filtering algorithm, can be transplanted to any microcontroller or other project code. In DSP28335, through verification.)
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Author: kirkwk
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Description: 本文从高速数字信号处理器的特点、自适应滤波器的原理及主要应用领域入手，介绍了自适应滤波器的基本理论思想，具体阐述了自适应滤波器的基本原理、算法及设计方法。本文中，对两种最基本的自适应算法，即最小均方误差(LMS)算法和递归最小二乘（RLS)算法进行了详细的介绍和分析，并针对两种算法的优缺点进行了详细的比较。最后用DSP实现了自适应滤波器。实验结果表明，该自适应滤波器滤波效果优越。(Starting from the characteristics of high-speed digital signal processor, the principle of adaptive filter and the main application fields, the basic theory of adaptive filter is introduced, and the basic principle, algorithm and design method of adaptive filter are described in detail. In this paper, two basic adaptive algorithms, namely the minimum mean square error (LMS) algorithm and recursive least squares (RLS) algorithm are introduced and analyzed in detail, and the advantages and disadvantages of the two algorithms are compared in detail. Finally, the adaptive filter is implemented by DSP. The experimental results show that the adaptive filter has better filtering effect.)
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Author: 古古怪怪asd
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