一种块处理模式数字滤波器的DSP实现

 对称FIR(有限冲激响应)滤波器的严格的线性相位特性和系统稳定性在数据传输、图像处理等领域得到了广泛的应用。基于单点处理模式的传统数字滤波器能实时处理点信号,但在处理图像、语音等帧信号时,速度较慢,实时性不强。针对这个问题提出了块处理模式的解决方案。通过分析FIR滤波器的原理,设计出基于块处理模式的对称FIR滤波器,并在TMS320C5510芯片中实现。介绍了基于双指针循环寻址的滤波算法,并给出了相应的主要程序。计算机试验仿真表明块处理模式方案提高了帧信号的处理速度,滤波性能良好。     

基于无限脉冲响应滤波器的自适应滤波E型有源噪声控制算法

相对于有限脉冲响应(FIR)滤波器,无限脉冲响应(ⅡR)滤波器由于其本身自有的零-极点结构,能以较低的阶数即可与系统模型匹配。因此在有源噪声控制(ANC)中,使用ⅡR滤波器可以大大节省计算量并能提高系统效率。自适应滤波U型最小均方差(FULMS)算法是目前常用的基于ⅡR滤波器的有源噪声控制算法,但该算法却不能保证全局收敛,这大大限制了该算法在有源噪声控制中的应用。本文提出一种新的基于ⅡR滤波器的自适应有源噪声控制算法——滤波E型最小均方差算法,该算法计算量相对于FULMS算法仅略有增加,但具有良好的全局收敛性,仿真结果表明该算法具有良好的降噪效果。     

波前功率谱密度(PSD)测量滤波器的设计

在采用波前功率谱密度评价大口径光学元件时，由于测得的原始数据受到测量系统引入的各种噪音和外界干扰的作用，会使检测准确度下降.对比引入带通滤波技术，对原始数据的有效滤波，减小带外的噪音分量和频谱混叠现象的影响.分别设计了无限脉冲响应带通滤波器和有限脉冲响应带通滤波器，通过模拟分析比较了两者的性能，得出在波前波前功率谱密度用于评价大口径光学元件的面形质量时，采用有限脉冲响应滤波器可以得到更高的测量准确度.     

基于DSP的全数字低场MRI信号接收算法研究

针对低场磁共振成像（MRI）系统,提出一种实用的基于通用浮点DSP的全数字高精度、高效率信号接收算法. 对经前端调理的回波信号直接采样数字化,在通用浮点DSP上进行数字正交解调, 采用高效的积分梳状滤波器(CIC)、半带滤波器(HB)、FIR滤波器等进行多级滤波抽取,极大的减少了运算量,提高了运算效率,最后得到用于成像的原始数据. 整个数字信号处理过程全部采用单精度浮点数据格式,用软件实现,提高了运算精度,增加了信号检测系统的灵活性. 最后通过CCS2.2软件仿真,验证了本算法的实时性和实用性.     

机械抖动激光陀螺新型信号处理方法的研究

针对高速采样低通滤波的线性脉冲响应(FIR)滤波器在机械抖动激光陀螺(MDRLG)信号处理中存在着较大的时延,限制MDRLG在快速跟踪中的应用,基于相关滤波技术提出了一种新的信号处理方法。对这一方法的基本理论进行了分析,并基于Labview实现了仿真。相关滤波后,激光陀螺输出中的抖动信号衰减了约41dB,还不能达到激光陀螺应用的要求,可以适当加一低阶的FIR低通滤波器去除抖动剥除后剩余的抖动成分和高频噪声。在静态条件下用这三种信号处理方法分别对激光陀螺零偏进行了测试。相关滤波法所测得零偏稳定性(1σ00)比50阶FIR滤波器差了将近三倍,相关滤波后再经过22阶的FIR低通滤波测得的零偏稳定性比50阶FIR低通滤波器的测试结果大了不足0.001°/h,延迟时间却从4.9ms减小至2.1ms。     

三维场景相息图的白光再现研究

讨论了相息图白光再现的可行性,得到采用在相息图和观察者之间加入狭缝的方法可实现相息图的白光再现的结论.提出了一种通过对彩虹全息图进行复数滤波来制作相息图的方法.根据彩虹全息图中物光的频谱宽度设计有限脉冲响应数字滤波器.对该滤波器进行移频,得到与通带位置物光的频谱位置一致的复系数带通滤波器.对全息图进行空域滤波,去除共轭像和零级,求出调制后物光的位相分布,制作场景相息图.实验观察到了经过此方法制作的相息图在白光下的再现像,表明本文的理论分析正确,所提出的方法可行.     

高速相干光通信系统中残留色散均衡器研究

在高速相干光通信系统中,完全使用电子色散补偿时会引入均衡增强相位噪声.本文提出在已有10Gbit/s光纤通信链路上使用电子色散补偿来补偿残留色散的混合色散补偿方案,实现了特定通道100Gbit/s以上高速数据传输.使用虚拟光学仪器软件仿真了在112Gbit/s相干光通信系统中,分别采用有限脉冲响应均衡器和重叠频域均衡器对现有光纤信道进行残留色散补偿后的系统性能.仿真结果表明:相同的抽头长度下,有限脉冲响应均衡器相比重叠频域均衡器有更好的残留色散补偿能力,并且有限脉冲响应均衡器均衡性能更稳定而且计算复杂度更低.因而,有限脉冲响应均衡器相较于重叠频域均衡器更适于混合色散补偿.     

基于FPGA的点目标滤波检测算法

介绍和分析了在当前成像跟踪系统中常用的几种点目标滤波检测算法。为了满足图像处理实时性要求，设计了一套针对图像滤波算法的FPGA硬件实现结构。该结构具有FPGA高速并行计算能力，能在信号读出的过程中实时地完成多种滤波处理。成像实验证明该方案切实可行，具有良好的实时滤波效果。     

一种基于数字信号处理技术的改进通用色散介质移位算子时域有限差分方法

借鉴数字信号处理技术中的无限脉冲响应滤波器实现的思想,提出一种适用于通用色散介质的改进移位算子时域有限差分（shift operator finite-difference time-domain,SO-FDTD）方法.与原SO-FDTD方法相比,改进的SO-FDTD方法内存占用减少33%以上,同时计算时间也有所减少.最后通过一阶、二阶色散介质雷达散射截面计算验证了方法的通用性和有效性. 关键词： 时域有限差分 色散介质 移位算子     

基于有限脉冲响应滤波器的数字全息零级像消除

针对数字全息中零级像的存在影响数字再现像的质量,分析数字全息图的记录、再现原理及频谱特性,提出了一种利用有限脉冲响应滤波器消除全息图数字再现中零级像的方法.该方法只需记录一幅数字全息图,不需要相移器材或其他辅助设备,直接利用数字图像处理对数字全息图在空域进行预处理,消除全息再现时的零级像干扰.对比了数字模拟和实验拍摄到的全息图在应用有限脉冲响应滤波器消除零级像前后的再现结果,表明该方法可消除零级衍射像,改善重建像质量,并且算法简单.     

Lattice form adaptive infinite impulse response filtering algorithm for active noise control

In some situations of active noise control, infinite impulse response (IIR) filters are more suitable than finite impulse response (FIR) filters owing to the poles in the transfer function. A number of algorithms have been derived for applying IIR filters in active noise control; however, most of them use the direct form IIR filter structure, which faces the difficulties of checking stability and relatively slow convergence speed for noise composed of narrow-band components with large power disparity. To overcome these difficulties along with using the direct form IIR filters, a new adaptive algorithm is proposed in this paper, which uses and updates the lattice form adaptive IIR filter in an active noise control system. Full mathematical derivations of the proposed algorithm are presented, and the comparison between the proposed algorithm and the commonly used filtered-u LMS and filtered-v LMS algorithms shows the superiority of the proposed algorithm.     

Adaptive Laguerre filters for active noise control

Feedforward controllers are used in many active noise control (ANC) systems to generate destructive interference in noise fields. An ideal feedforward ANC controller should have an infinite impulse response (IIR) transfer function, but most available feedforward ANC controllers have finite impulse responses (FIR) instead. The main reason is related to the adaptation algorithms of ANC systems. In general, adaptive FIR filters converge faster with guaranteed stability. In this study, the adaptive Laguerre filter is proposed and tested in an ANC application with positive experimental effects. The new ANC controller is an IIR filter, but its adaptation is similar to that of a FIR filter with fast convergence and guaranteed stability. Detailed explanations and analysis are presented in the main text.     

机车驾驶室中自适应有源降噪系统的研究

根据机车驾驶室中噪声特征的分析结果，提出了采用FIR滤波器的自适应有源降噪(ANNC)系统。为消除机车驾驶室中严重的背景噪声干扰影响，应用伪随机相关法对驾驶室中的噪声脉冲响应进行测量。通过数字仿真研究，对基于IMS算法的ANNC系统的关键参数进行了寻优。最后，在实际声场中建立了实时双通道ANNC系统并进行了相关的实验验证。仿真与实验的结果证明了本文提出的AANC系统的可行性和有效性。     

Efficient array beam forming by spatial filtering for ultrasound B-mode imaging

This paper proposes an efficient array beam-forming method using spatial matched filtering (SMF) for ultrasonic imaging. In the proposed method, ultrasonic waves are transmitted from an array subaperture with fixed transmit focus as in conventional array imaging. At receive, radio frequency echo signals from each receive channel are passed through a spatial matched filter that is constructed based on the system transmit-receive spatial impulse response. The filtered echo signals are then summed without time delays. The filter concentrates and spatially registers the echo energy from each element so that the pulse-echo impulse response of the summed output is focused with acceptably low side lobes. Analytical beam pattern analysis and simulation results using a linear array show that this spatial filtering method can improve lateral resolution and contrast-to-noise ratio as compared with conventional dynamic receive focusing (DRF) methods. Experimental results with a linear array are consistent but point out the need to address additional practical issues. Spatial filtering is equivalent to synthetic aperture methods that dynamically focus on both transmit and receive throughout the field of view. In one common example of phase aberrations, the SMF method was degraded to a degree comparable to conventional DRF methods.     

IMPLEMENTATION OF A BROADBAND DUCT ANC SYSTEM USING ADAPTIVE SPATIALLY FEEDFORWARD STRUCTURE

An adaptive spatially feedforward algorithm is proposed for broadband attenuation of noise in ducts. Acoustic feedback generally exists in this active noise control structure. Munjal and Eriksson (1988 Journal of Acoustical Society of America84, 1086-1093) derived an ideal controller for the spatially feedforward structure. The ideal controller can be partitioned into two parts. The first part represents a repetitive controller that can be implemented by an infinite impulse response (IIR) filter, whereas the second part represents the dynamics of transducer that can be implemented by a finite impulse response (FIR) filter. In the paper, the IIR filter is merged with the original plant. The FIR filter is adaptively updated by the least-mean-square (LMS) algorithm to accommodate perturbations and uncertainties in the system. The proposed algorithm is implemented via a floating point digital signal processor and compared with other commonly used algorithms such as the Filtered-X LMS algorithm, the feedback neutralization algorithm, and the Filtered-U LMS algorithm. Experimental results show that the system has attained 15·7 dB maximal attenuation in the frequency band 200-600 Hz.     

Background noise cancellation of manatee vocalizations using an adaptive line enhancer

The West Indian manatee (Trichechus manatus latirostris) has become an endangered species partly because of an increase in the number of collisions with boats. A device to alert boaters of the presence of manatees is desired. Previous research has shown that background noise limits the manatee vocalization detection range (which is critical for practical implementation). By improving the signal-to-noise ratio of the measured manatee vocalization signal, it is possible to extend the detection range. The finite impulse response (FIR) structure of the adaptive line enhancer (ALE) can detect and track narrow-band signals buried in broadband noise. In this paper, a constrained infinite impulse response (IIR) ALE, called a feedback ALE (FALE), is implemented to reduce the background noise. In addition, a bandpass filter is used as a baseline for comparison. A library consisting of 100 manatee calls spanning ten different signal categories is used to evaluate the performance of the bandpass filter, FIR-ALE, and FALE. The results show that the FALE is capable of reducing background noise by about 6.0 and 21.4 dB better than that of the FIR-ALE and bandpass filter, respectively, when the signal-to-noise ratio (SNR) of the original manatee call is -5 dB.     

New demodulation filter in digital phase rotation beamforming

In this paper, we present a new quadrature demodulation filter to reduce hardware complexity in digital phase rotation beamforming. Due to its low sensitivity to phase delay errors, digital quadrature demodulation is commonly used in ultrasound machines. However, since it requires two lowpass filters for each channel to remove harmonics, the direct use of conventional finite impulse response (FIR) filters in ultrasound machines is computationally expensive and burdensome. In our new method, an efficient multi-stage uniform coefficient (MSUC) filter is utilized to remove harmonic components in phase rotation beamforming. In comparison with the directly implemented FIR (DI-FIR) and the previously-proposed signed-power-of-two FIR (SPOT-FIR) lowpass filters, the proposed MSUC filter reduces the necessary hardware resources by 93.9% and 83.9%, respectively. In simulation, the MSUC filter shows a negligible degradation in image quality. The proposed method resulted in comparable spatial and contrast resolution to the DI-FIR approach in the phantom study. These preliminary results indicate that the proposed quadrature demodulation filtering method could significantly reduce the hardware complexity in phase rotation beamforming while maintaining comparable image quality.     

Measuring short impulse responses with inverse filtered maximum-length sequences

Most loudspeakers have a non-flat frequency response which produces a long oscillating impulse response. An inverse filtering approach may be used to calculate the driving waveform necessary to equalize the response of the loudspeaker in order to radiate shorter acoustic pulses. When combined with the MLS technique, inverse filtering may be used to pre-emphasize the driving signal so that a shorter impulse response, with a prescribed waveform, is measured. This technique is described and illustrated by applying it to a distributed mode loudspeaker. Originally, this loudspeaker has a rather irregular response in a wide band. When the MLS signal is pre-emphasized with the proper inverse filter, a shorter impulse response is measured with a zero-phase cosine-magnitude spectrum.