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自适应旁瓣对消在雷达中的应用 总被引:6,自引:1,他引:6
自适应旁瓣对消(ASLC)是雷达抗有源干扰的有效方法.它采用空间滤波技术,通过辅助接收通道在干扰方向形成波束图的零点,实现对干扰信号的抑制.本文介绍了自适应旁瓣对消的原理,然后给出了基于DSP的ASLC实现方案,分析了自适应旁瓣对消的性能.同传统的对消系统相比,该系统具有更好的抗干扰性能. 相似文献
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在雷达检测接收机之输入端的目标信号的平均功率与不需要的信号的平均功率之比,通过采用某些基于利用回波极化状态的自适应信号处理方法,而有可能大大提高。不需要的雷达回波的抑制通常是通过一个双极化雷达信号的矢量处理而获得的,这等效于使接收天线的状态自适应于不需要的环境状况。在本文中,在讨论某些关于部分极化的不需要的雷达回波的自适应对消概念之后,分析一种新的非线性自适应极化对消器的工作与性能。这种对消器在存在阻塞干扰或降雨杂波情况下可用于改善信号一干扰比。在文末,展示了某些与实验双极化雷达数据有关的结果。 相似文献
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分析了参考信号耦合有用信号时,实际自适应干扰对消系统的时域特性。分析表明,参考信号含有用信号导致系统时变,改变系统的最佳权值,使系统的干扰对消效果下降,并对有用信号的接收产生影响。根据实际共平台通信系统的收发耦合特性,从时域角度分析得到系统干扰对消比和有用对消比的计算表达式,并得到干扰对消比和有用对消比随相对时延相位变化的规律。提出一种时延匹配方法,通过匹配参考信号提取点至正交功分器输出与接收天线至合成信号注入点间的时延,可以有效抑制参考信号耦合有用信号导致干扰对消效果下降的问题。仿真结果表明,所提出的时延匹配方法能消除有用信号对干扰对消比的影响,还能提高干扰对消效果。 相似文献
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提出了一种基于改进变步长自适应滤波的密集假目标干扰抑制方法。该方法首先采用stretch去斜处理,消去信号的二次项,并在频域上分离目标回波与干扰,从而得到频率不同的信号分量;然后用基于类箕舌线函数的变步长自适应滤波方法,对stretch处理后的回波信号滤波;最后对消密集假目标干扰,将对消后的信号作逆stretch处理恢复目标回波。所提出的类箕舌线函数调节自适应滤波方法中的步长,能有效地抑制距离目标回波较近的假目标干扰。仿真结果表明,改进变步长自适应滤波方法在目标回波与干扰时延差较小时,也可有效地对回波信号滤波,干扰对消效果明显;与基于定步长的滤波方法相比,所提方法在信干比较低的环境下仍具有良好的鲁棒性,并能有效抑制密集假目标干扰。 相似文献
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宽带泄漏信号自适应对消器的性能 总被引:1,自引:0,他引:1
自适应对消器可以有效地消除雷达发射机宽带泄漏信号对雷达系统本身的影响。本文依据环路模型建立了描述自适应对消器工作过程的随机微分方程,求得了随机微分方程的瞬态解及其统计特性。最后导出了衡量对消器性能的干扰对消比的理论公式。 相似文献
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自适应波束形成(ADBF)和自适应旁瓣对消算法都能够很好地抑制旁瓣干扰,但是由于干扰协方差矩阵的影响,ADBF在抑制干扰同时给和、差单脉冲测角带来误差.文中针对自适应旁瓣对消算法,系统研究了其在和、差单脉冲测角中的应用,着重分析了辅助天线中的目标信号对测角性能的影响.理论分析和仿真结果表明,辅助天线中的目标信号对测角的影响主要存在于辅助天线的增益大于差波束增益的角域,而在其他角域,影响很小. 相似文献
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An "event-synchronous interference canceller" (ESC) for cancellation of electrocardiographic (ECG) interference in diaphragmatic electromyographic (EMGdi) signals is addressed in this paper. ESC pursues the concept of the "event synchronous adaptive interference canceller" (ESAIC), which was proposed in [1] as a specific application of the well known "adaptive noise canceller" (ANC) paradigm, but ESC uses a simple adaptive gain control (AGC) instead of the complex adaptive filter of the ANC. The proposed ESC method is evaluated using both computer simulations and real EMGdi data, and its efficiency in interference cancellation is compared to that of ESAIC. Of particular interest is the result that the ESC can replace the ESAIC providing better performance as well as a considerable reduction of computational costs. 相似文献
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Kohno R. Imai H. Hatori M. Pasupathy S. 《Selected Areas in Communications, IEEE Journal on》1990,8(4):691-699
The authors propose and investigate an adaptive canceller of intersymbol and cochannel interference due to channel distortion and cross-correlation among pseudonoise sequences assigned to individual users of a DS-SSMA (direct-sequence spread-spectrum multiple-access) system. In order to implement a local area network (LAN) by using a power line installed in a building wall as a transmission channel, the authors have investigated utilization of DS-SSMA which has advantages such as robustness against narrow-band interference and noise and realization of asynchronous code division multiple access. In a power line, however, restriction of transmission bandwidth for communications makes it difficult to suppress cochannel interference and the channel is also time-varying due to fluctuation of loads. Since the proposed canceller adaptively eliminates cochannel interference as well as intersymbol interference, it can facilitate synchronization and increase the number of the simultaneously accessing users on a power line with restricted processing gain. The error probability in the output of the canceller is theoretically calculated for the steady-state case by using a Markov model. Computer simulations illustrate stable convergence properties of the canceller 相似文献
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利用小波神经网络作为噪声对消滤波器,实现了参考噪声与干扰噪声呈非线性相关条件下的噪声对消。在参考噪声与干扰噪声非线性相关时,传统的横向滤波器效果不理想,利用小波神经网络的非线性特性,可更好的解决非线性噪声条件下的噪声对消问题。计算机仿真结果证明,小波神经网络噪声对消在非线性噪声条件可有效提高信噪比增益。 相似文献
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噪声对消在信号处理系统中的应用 总被引:3,自引:1,他引:2
论述基于LMS算法的自适应滤波器噪声对消的工作原理,以及基于AR模型的信号分析方法。在这两种方法相结合的情况下,能有效去除信号的噪声。对含有瞬态干扰的微弱信号,用AR模型法估计出噪声的系数并预测噪声波形,通过自适应滤波器噪声对消原理进行滤波,最后在Matlab环境下进行仿真试验,结果表明该方法具有较好的去噪效果。 相似文献
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传统的自适应滤波去噪的系统中需要一个参考信号,实际中一般事先并不知道噪声的特性,而且噪声也是时变的。因此提出一种新的基于盲噪声抑制方法,可降噪并跟踪噪声特性的变化,随时调节滤波器的系数,比传统的自适应噪声控制具有优势。 相似文献
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二元麦克风小阵列在手机、助听器等受空间、成本以及运算能力限制的设备中被广泛研究用以提高目标语音质量。二元麦克风小阵列中语音增强算法主要包括波束形成方法以及相干性滤波器方法。波束形成方法的思想是利用目标声源相对阵列的位置关系获取相应的时域和空域信息,可以保留目标声源方向的信号而抑制其他方向的干扰信号;相干性滤波器方法则通过阵元间不同信号的相关性进行噪音抑制。考虑这两种类型方法的优点,本文提出一种面向二元麦克风小阵列改进的广义旁瓣抵消器语音增强算法,通过在广义旁瓣抵消器的固定波束形成支路上使用相干性滤波器,提高固定波束形成输出信号的信噪比,然后在广义旁瓣抵消器自适应支路利用阵列的时域和空域信息对固定波束形成支路输出的信号中残余噪音进行估计,进而获得增强后目标输出信号。仿真和实际试验表明,本文提出的算法明显优于单独使用小阵列波束形成算法和相干性滤波器算法。 相似文献
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This paper proposes a new robust adaptive beamformer applicable to microphone arrays. The proposed beamformer is a generalized sidelobe canceller (GSC) with a new adaptive blocking matrix using coefficient-constrained adaptive filters (CCAFs) and a multiple-input canceller with norm-constrained adaptive filters (NCAFs). The CCAFs minimize leakage of the target-signal into the interference path of the GSC. Each coefficient of the CCAFs is constrained to avoid mistracking. The input signal to all the CCAFs is the output of a fixed beamformer. In the multiple-input canceller, the NCAFs prevent undesirable target-signal cancellation when the target-signal minimization at the blocking matrix is incomplete. The proposed beamformer is shown to be robust to target-direction errors as large as 200 with almost no degradation in interference-reduction performance, and it can be implemented with several microphones. The maximum allowable target-direction error can be specified by the user. Simulated anechoic experiments demonstrate that the proposed beamformer cancels interference by over 30 dB. Simulation with real acoustic data captured in a room with 0.3-s reverberation time shows that the noise is suppressed by 19 dB. In subjective evaluation, the proposed beamformer obtains 3.8 on a five-point mean opinion score scale, which is 1.0 point higher than the conventional robust beamformer 相似文献
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An adaptive polyspectral canceller configuration is defined, whereby auxiliary canceller channels are formed using distinct monomial expressions of the auxiliary sensor channels. If noises in the auxiliary canceller channels are correlated with the noise in a main channel, then improvement in output signal-to-noise (S/N) power ratio is possible by cancelling in linear fashion the correlated auxiliary canceller channels with the main channel. The convergence performance of the polyspectral canceller is analyzed. A simple expression is derived for the asymptotic S/N efficiency of the adaptive polyspectral canceller as a function of the number of independent input sample vectors used to calculate the adaptive canceller weights and other canceller/noise model parameters. It is shown by simulation for low-order polyspectral cancellers of a specific form with Gaussian inputs that this asymptotic expression is a good approximation of the actual S/N efficiency for a moderate number of input sample vectors. However, for moderate-to-high order polyspectral cancellers, the asymptotic expression is a poor indicator of performance 相似文献
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Miao Jialin 《电子科学学刊(英文版)》1985,2(1):49-58
A modified time domain orthogonality algorithm (MTDO) for adaptive filters is presented and described. The results of computer
simulation show that the algorithm has two characteristics: 1. the same convergence factorc can be used within a fairly wide range of signal to noise ratio (SNR) of input signal. 2. MTDO adaptive noise canceller has
a useful characteristic, i. e. at the same signal to noise ratio of input desired signal the input reference signal power
is uncorrelated with the convergence factorc. Therefore one adaptive noise canceller can be used more flexibly. The performance of spectral estimates of the MTDO adaptive
system is obtained by us experimentally and compared with the least mean square algorithm adaptive system. 相似文献