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新型800MHz发射机滤波器设计 总被引:6,自引:0,他引:6
以800MHz带通滤波器(BPF)为例,论述了如何应用Bessel函数进行发射机BPF的设计,同时利用Pspice仿真软件对设计结果进行波特图和群延迟特性仿真.用最优化理论对所设计出的发射机滤波器进行调整,使其性能误差最小不超过0.2%.在工程应用上与其他发射机滤波器进行比较,表明具有应用价值,可用于无线广播、移动通信等领域. 相似文献
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分数延迟滤波器广泛用于通信,语音处理,回声消除等。该文基于信号的多分辨空间一般模型,由尺度函数给出最优FIR分数延迟滤波器的设计方法,并进行数值试验证实其有效性。 相似文献
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新型微带发夹型双通带滤波器的设计 总被引:2,自引:1,他引:1
提出一种新型微带发夹型双通带滤波器,给出滤波器的设计原理及结构,分析比较不同结构参数对滤波器特性的影响,并通过仿真优化得出其特性曲线图.结果表明,设计的微带双通带滤波器拥有良好的通带特性,实现了2.4GHz/5.2GHz 和2.4GHz/5.8GHz 的双通带.同时,文中设计的滤波器的尺寸很小,便于实现系统的小型化. 相似文献
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通过合理选择滤波器的类型和阶数,利用低Q值二级滤波器放在高Q值二级滤波器的前级,给每一级分配不同的增益使得每一级的输出峰值相同.从开关电容电路的原理入手,分析了开关电容电路和电容编程阵列,最终设计一个可编程开关电容6阶带通滤波器.选择合适的运算放大器参数.可编程滤波器系统共需3路时钟控制,滤波器编程参数控制模块用于实现芯片内部程序存储器编程控制.通过对设计的开关电容滤波器进行仿真,结果基本与设计目标吻合. 相似文献
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传统的交指带通滤波器设计采用公式计算归一化自电容和互电容,并通过查表计算得到滤波器的参数,该方法计算繁琐,且计算误差较大.文中介绍了一种根据谐振器的直接耦合原理,利用两终端的外部品质因数Q和谐振器间的耦合系数K设计抽头式交指型带通滤波器的方法.据此,设计了一种用于微波选频的交指滤波器,借助ADS软件对滤波器进行仿真和优化.通过仿真及实测结果的分析对比,进一步给出了设计改进方案. 相似文献
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线谱对参数具有优越的量化特性和内插特性,以线谱对参数作为滤波器系数的LSF滤波器是一种适用于高压缩率声码器的声道滤波器.MATALAB功能强大且编程方便,广泛应用于各种算法的性能评估.文章简要介绍了LSF声道滤波器原理,给出了用MATLAB实现LSF滤波器的流程图和程序代码.最后设计了一个LSF声道滤波器的应用实例,并对应用实例进行了MATLAB仿真,给出了应用实例的仿真结果. 相似文献
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Ka频段薄膜滤波器设计 总被引:1,自引:0,他引:1
介绍了一种简单有效的Ka频段薄膜微带滤波器设计方法。通过选
择恰当的滤波器模型,提取参数和初值,用ADS和Designer两种仿真软件结合进行设计,得
到了理想的滤波器响应曲线。通过三轮滤波器投版测试得到工艺补偿准确值,用于修正仿真
设计和滤波器实际曲线之间的偏差,最后达到了投片测试结果和仿真设计基本吻合的目的。 相似文献
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本文建议一种多维递归数字滤波器的并行算法,其于常规次第算法相比,不造成任何附加算术运算;并推导出在全并行处理时滤波器系数所需满足的条件。对角线信号处理法是本文所述方法的一个重要的特例,故对对角线采样数据的频谱进行了研究。本文建议的并行算法可以分别用向量处理器和专用硬件来实现;后一种情况,既可使用全并行也可使用部分并行。 相似文献
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对于传统的固定系数滤波器,分布式算法可以利用查找表有效提高其运行速度;但是对于自适应滤波器,其系数是不断调整的,不可以直接应用分布式算法。依据分布式算法设计了一种适用于自适应滤波器的现场可编程门阵列(FPGA)实现方法,并在Xilinx公司提供的ISE软件平台上应用Verilog硬件设计语言(HDL)进行编程,利用Modelsim和Matlab软件进行了仿真。仿真结果基本一致,验证了自适应有限冲激响应滤波器的FPGA实现方法是可行的。 相似文献
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The most computationally intensive part of the wideband receiver of a software defined radio (SDR) is the intermediate frequency (IF) processing block. Digital filtering is the main task in IF processing. The computational complexity of finite impulse response (FIR) filters used in the IF processing block is dominated by the number of adders (subtracters) employed in the multipliers. This paper presents a method to implement FIR filters for SDR receivers using minimum number of adders. We use an arithmetic scheme, known as pseudo floating-point (PFP) representation to encode the filter coefficients. By employing a span reduction technique, we show that the filter coefficients can be coded using considerably fewer bits than conventional 24-bit and 16-bit fixed-point filters. Simulation results show that the magnitude responses of the filters coded in PFP meet the attenuation requirements of wireless communication standard specifications. The proposed method offers average reductions of 40% in the number of adders and 80% in the number of full adders needed for the coefficient multipliers over conventional FIR filter implementation methods 相似文献
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We propose a new bandpass filter (BPF)‐based online channel normalization method to dynamically suppress channel distortion when the speech and channel noise components are unknown. In this method, an adaptive modulation frequency filter is used to perform channel normalization, whereas conventional modulation filtering methods apply the same filter form to each utterance. In this paper, we only normalize the two mel frequency cepstral coefficients (C0 and C1) with large dynamic ranges; the computational complexity is thus decreased, and channel normalization accuracy is improved. Additionally, to update the filter weights dynamically, we normalize the learning rates using the dimensional power of each frame. Our speech recognition experiments using the proposed BPF‐based blind channel normalization method show that this approach effectively removes channel distortion and results in only a minor decline in accuracy when online channel normalization processing is used instead of batch processing. 相似文献
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Time domain processing of broadband signals using an array of sensors is normally carried out by a transversal filter, and the filter coefficients are adjusted by solving a constrained beamforming problem that requires manipulation of matrices of dimension LJ×LJ with L denoting the number of sensors in the array and J denoting the filter length. This paper studies a frequency domain method to compute the coefficients of transversal filter, compares its performance, sensitivity to the look direction, and computational efficiency with that of the direct method, and shows how the computational saving increases using the frequency domain method as the number of elements and the filter length are increased. The paper presents a number of techniques that exploit the inherent structure present in the frequency domain method to reduce the computation load and shows that the real-time computational saving of the order of 12520 folds is possible for an array of 100 elements using a tap delay line filter of 100 taps. The paper demonstrates that a beamformer using this method is able to produce more output SNR with less computation time than the one using the normal time domain method 相似文献
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FIR数字滤波器广泛应用于实时数字信号处理领域.本文介绍了FIR数字滤波器的结构、特点及设计方法,并采用窗函数法设汁了FIR滤波器.利用TMS320VC5509 DSP芯片强大的数字信号处理功能实现了该滤波器.实验表明,此数字滤波器工作稳定,能够满足实时的滤波处理功能. 相似文献
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在传统图像增强技术的基础上,提出了一种基于GABOR小波系数融合的图像增强方法。通过Gabor小波对图像进行滤波,产生八个方向的滤波图像系数,对这些图像系数用求平均值融合方法得到图像边缘图,再融合经传统空域增强方法得到的直方图均衡化图像以产生最终增强的边缘信息以便于分析处理的图像。 相似文献
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Heung Mook Kim Sung Ik Park Jae Hyun Seo Homin Eum Yong-Tae Lee Soo In Lee Hyuckjae Lee 《Broadcasting, IEEE Transactions on》2008,54(2):249-256
This paper presents novel modulation and pre-equalization methods to minimize a signal processing time delay in the equalization digital on-channel repeater (EDOCR) for the ATSC terrestrial digital TV system. The proposed modulation method uses equi-ripple (ER) filter for vestigial side bands (VSB) pulse shaping instead of conventional square root raised cosine (SRRC) filter. And the proposed pre-equalization method calculates pre-equalizer filter coefficients by comparing a baseband signal as a reference signal and a demodulated repeater output signal, and then creates new VSB pulse shaping filter coefficients by the convolution of the ER filter and the pre-equalizer filter coefficients. The new VSB pulse shaping filter minimizes the time delay of EDOCR by adjusting the number of its pre-taps and also compensates the linear distortions due to the use of the ER filter and mask filter. 相似文献