基于FD216芯片的自适应均衡器的实现

描述了基于FD216数字信号处理器的LMS自适应均衡器系统.该系统是在FD21 6 EZ-LABEvaluation Board上实现的,充分发挥了芯片的专用硬件逻辑、专业化的指令集作用,有效快速地完成了仿真ISI的自适应均衡,突出反映了FD21 6芯片的高速数字信号的处理能力.     

数字阵列雷达通道自适应均衡器的一种设计方法

通道之间的幅相不一致性和通道带内频响的不一致性严重影响了数字阵列雷达的性能,采用自适应均衡技术可以解决通道的一致性问题。高性能的数字信号处理芯片可以在较高的运算速度下,实现通道的自适应均衡。本文采用最小二乘广义求逆法求解均衡器系数,提出了利用多通道下变频芯片中的可编程滤波器进行均衡的方法。最后通过仿真和测试结果,分析了均衡性能。     

基于TMS320C50的通信保密机硬件设计

介绍了以数字信号处理器芯片TMS320C50为核心的保密机实现方案。围绕C50这一中心将系统分为C50及外围逻辑电路、存储器、TLC32044及外围电路和电话接口电路4部分硬件设计。该方案采用一片TMS320C50即可完成所有的加解密、回波抵消运算，并为密钥控制、自适应均衡等运算保留了系统资源。     

基于DSP的自适应回波抵消系统的设计

能通信保密机课题为背景,介绍以数字信号处理器芯片TMS320C50为核心的保密机实现方案。该方案采用一片TMS320C50即可完成所有的加解密、回波抵消运算,并为密钥控制、自适应均衡等运算保留了系统资源。     

绞合线对上工作的全双工数字收发机模拟前端

已经研制成适合回波对消的绞合线对上工作的全双工数字收发机模拟前端CMOS芯片。这个芯片包括发射、接收、平衡、均衡等滤波器以及12位6μs的自校准A/D转换器。此芯片与数字信号信息处理机接口,执行收发机其余的功能,例如回波对消,自适应平衡,定时恢复等。     

G.723.1语音编解码在FD216上的实现

介绍了ITU-T G.72 3.1标准和FD216的特征.分析了以FD216实现编解码系统的设计要点,设计了G.72 3.1实时语音编解码系统.该系统实现了ITU-T G.72 3.1标准包括附录A在内的全部功能,通过了所有测试矢量.     

有源噪声控制的声学和硬件／软件考虑

在源噪声是自适应滤波器的实时应用，它要求用现代快速运算数字信号处理的全部功能来实现，即使有用频率范围很低，也需要用数千赫的采样频率，算法要求有效的高速处理，系统硬件设计允许软件的多功能控制以及有源降噪系统的自动操作，本文给出了有源噪声控制的声学和硬件／软件考虑，讨论了使用ＡＤＳＰ－２１０１／２１０５芯片实现有源降噪自适应控制。     

基于FPGA的自适应均衡器的研究与设计

近年来,自适应均衡技术在通信系统中的应用日益广泛,利用自适应均衡技术在多径环境中可以有效地提高数字接收机的性能.为了适应宽带数字接收机的高速率特点,本文阐述了自适应均衡器的原理并对其进行改进.最后使用FPGA芯片和Verilog HDL设计实现了自适应均衡器并仿真验证了新方法的有效性.     

基于DSP5509A的自适应系统识别的实现

在数字信号处理中,自适应系统识别是处理一些未知系统的重要方法。文中在论述自适应滤波器的基本原理和LMS算法的基础上,阐述了自适应系统识别的实现原理。然后在DSP的集成开发环境下设计一个随机白噪声信号发生器,用来产生自适应滤波器和未知系统所需的测试数据。最后采用FIR滤波器作为待识别的未知系统,并利用TI公司的DSP TMS320VC5509A芯片实现了自适应系统识别。实验结果表明,该方法实现比较简单,且能达到信号处理的高精度和高效性。     

一种新型矿吉中频感应数字移动通信系统的探讨

文章针对井下特殊的通信环境、借鉴数字移动通信中自适应增量调制和以宽频带换取抗衰落，抗多径干扰的多进制数字调频技术确ＭＦＳＫ设备复杂，占用频带宽等缺点，提出了一种以高速数字信号处理芯片和大订成锁相频率合成电路为基础的新型通信系统。     

A step-by-step quasi-Newton algorithm in the frequency domain and its application to adaptive channel equalization

In this paper, a new efficient adaptive filtering algorithm belonging to the Quasi-Newton (QN) family is proposed. In the new algorithm, the involved inverse Hessian matrix is approximated by a proper expansion, consisting of powers of a Toeplitz matrix. Due to this formulation, the algorithm can be implemented in the frequency domain (FD) using the fast Fourier transform (FFT). Efficient recursive relations for the frequency domain quantities updated on a step-by-step basis have been derived. The proposed algorithm turns out to be particularly suitable for adaptive channel equalization in wireless burst transmission systems. Based on this approach, new adaptive linear equalization (LE) and decision feedback equalization (DFE) algorithms have been developed. These algorithms enjoy the combined advantages of QN formulation and FD implementation, exhibiting faster convergence rate than their stochastic gradient counterparts and less computational complexity, as compared with other Newton-type algorithms.     

Frequency-Domain Turbo Equalization and Multiuser Detection for DS-UWB Systems

One of the major challenges in direct sequence-ultra wideband (DS-UWB) receiver design is intersymbol interference (ISI). Several equalization schemes to eliminate ISI in DS-UWB systems have been proposed in the literature. It was shown that frequency-domain (FD) equalization techniques can offer better trade off between performance and complexity compared to timedomain equalization schemes for DS-UWB systems on highly dispersive channels. In this paper, we derive low-complexity FD minimum mean square error turbo equalization schemes for single-user binary phase shift keying (BPSK) and quaternary bi-orthogonal keying (4BOK) DS-UWB systems. For multiuser DS-UWB systems, we combine FD turbo equalization schemes with soft interference cancelation to obtain multiuser FD turbo detectors. The bit error rate performance gain due to turbo detection is shown to be significant, particularly for multiuser DS-UWB systems.     

A Microprocessor-Oriented Algorithm for Adaptive Equalization

Adaptive equalization based on a transversal filter is extremely important for reliable data transmission over the voiceband telephone lines. Unfortunately, implementation of rapidly converging and small residual distortion algorithms on the standard MOS microprocessors or standard-cell MOS VLSI chips is difficult because of their mathematical complexity. In this paper, an adaptive equalization algorithm is proposed which is suitable for cost-effective implementation on MOS microprocessors and standard-cell MOS VLSI chips. It is based on the same principle of delayed decision that was used successfully in delta modulation. It involves no multiplications or analog-to-digital conversion, but is relatively fast and provides a relatively low residual distortion. The proposed algorithm is simulated in both noise-free and noisy environments and is compared to two well-known classical algorithms.     

Adaptive equalization for TDMA digital mobile radio

Adaptive equalization for a TDMA (time-division multiple-access) digital cellular system is discussed. A survey of adaptive equalization techniques that includes their performance characteristics and limitations and their implementation complexity is presented. The design of adaptive equalization algorithms for a narrowband TDMA system is considered. It is concluded that, on the basis of implementation complexity and performance in the presence of multipath distortion and signal fading, MLSE (maximum-likelihood sequence estimation) and DFE (decision feedback equalization) are viable equalization methods for mobile radio     

Forward-error-control (FEC)-assisted adaptive equalization fordigital cellular mobile radio

An adaptive equalization technique using forward error control coding is described and evaluated. Computer simulation is used to compare the FEC-assisted adaptive equalization to conventional adaptive equalization. The system model employed is based on the proposed digital cellular communication standard for North America. The FEC-assisted adaptive equalization technique performs better than the conventional equalization method at low and moderate Doppler frequencies with the same level of redundancy in the transmitted data. For systems employing both FEC coding and adaptive equalization, the FEC-assisted adaptive equalization method successfully combines these two functions while using the redundancy required for only one of them. In other words, training required for adaptive equalization can be provided at no extra cost in terms of data redundancy in a system using FEC coding and vice versa     

自适应复信道均衡的一种新的神经网络方法

近年来,神经网络已经广泛地应用到许多信号处理问题中.自适应信道均衡是数字通信系统中的一个重要问题.在本文中我们提出一种基于复数赫布类型算法的自适应信道均衡器.计算机模拟表明,无论在线性还是非线性信道中,所提出的均衡器均表现出良好的性能,这为自适应复信道均衡提供了一种新的方法.     

Outage Performance of QAM Digital Radio Using Adaptive Equalization and Switched Space Diversity Reception

Recently, the ability of anti-fading measures to reduce the outage which occurs on digital microwave radio links has been the subject of considerable study. Ideal and nonideal adaptive equalization in both the time and frequency domain have been evaluated for nondiversity reception using various performance criteria. Ideal adaptive equalization and space diversity reception have been considered using a recently published model of multipath fading on space diversity radio channels. In this paper, we determine the outage of 16-QAM and 64-QAM digital radio systems using adaptive slope equalization, finite-tap decision feedback equalization, and switched space diversity reception. The outage is evaluated by computing the probability of occurrence of those channel realizations which cause the bit error rate to exceed a critical value. The dependence of the outage prediction on the equalization method and the outage defining criterion is investigated by also considering ideal adaptive equalization and a signal-to-distortion ratio, respectively.     

基于一种杂交学习算法的自适应复信道均衡技术

本文提出了一种基于多层前馈神经网络杂交学习算法的自适应复信道均衡的新方法。该学习算法用来训练一个输入、输出、权值和激活函数均为复数的神经网络。神经网络的训练利用了监督和非监督相结合的杂交技术,而权值的调整是基于TLS（total least square)准则进行的。计算机仿真结果表明,无论是在线性还是在非线性信道中,所提出的方法都表现出了很好的性能,这为自适应复信道均衡提供了一种新方法。     

A Supervised Constant Modulus Algorithm for Blind Equalization

Blind equalization is a technique for adaptive equalization of a communication channel without the aid of the usual training sequence. Although the Constant Modulus Algorithm (CMA) is one of the most popular adaptive blind equalization algorithms, it suffers from slow convergence rate. A novel enhanced blind equalization technique based on a supervised CMA (S-CMA) is proposed in this paper. The technique is employed to initialize the coefficients of a linear transversal equalizer (LTE) filter in order to provide a fast startup for blind training. It also presents a computational study and simulation results of this newly proposed algorithm compared to other CMA techniques such as conventional CMA, Normalized CMA (N-CMA) and Modified CMA (M-CMA). The simulation results have demonstrated that the proposed algorithm has considerably better performance than others.