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    • 25. 发明申请
    • ULTRA-WIDEBAND RADAR SYSTEM USING SUB-BAND CODED PULSES
    • 超声波雷达系统使用子带编码脉冲
    • US20050179585A1
    • 2005-08-18
    • US10851635
    • 2004-05-21
    • Michael WalkerDavid FurunoNaiel AskarGerald Rogerson
    • Michael WalkerDavid FurunoNaiel AskarGerald Rogerson
    • G01S20060101G01S7/28G01S13/00G01S13/02G01S13/28
    • G01S13/284G01S13/0209G01S13/282
    • Methods and apparatus are provided for radar systems using multiple pulses that are shorter than the expected range delay extent of the target to be imaged. In one implementation, a method for performing radar includes the steps of: transmitting a plurality of pulses, each pulse having a different center frequency and a time duration shorter than an expected range delay extent of a target, wherein a total bandwidth is defined by a bandwidth occupied by the plurality of pulses; receiving reflections of the plurality of pulses; and performing pulse compression on the received pulse reflections to generate a detection signal having a radar resolution approximately equivalent to the transmission and reception of a single pulse having the total bandwidth. In preferred form, the pulses comprise ultrawideband (UWB) pulses each occupying a sub-band of the overall system bandwidth.
    • 为使用比要成像的目标的预期范围延迟范围短的多个脉冲的雷达系统提供方法和装置。 在一个实施方式中,执行雷达的方法包括以下步骤:发送多个脉冲,每个脉冲具有与目标的预期范围延迟范围不同的中心频率和持续时间,其中总带宽由 多个脉冲占用的带宽; 接收多个脉冲的反射; 并对所接收的脉冲反射进行脉冲压缩,以生成雷达分辨率近似等于具有总带宽的单个脉冲的发送和接收的检测信号。 在优选形式中,脉冲包括每个占据整个系统带宽的子带的超宽带(UWB)脉冲。
    • 28. 发明授权
    • Method and circuit arrangement for speech level measurement in a speech signal processing system
    • 用于语音信号处理系统中语音电平测量的方法和电路装置
    • US06539350B1
    • 2003-03-25
    • US09442392
    • 1999-11-18
    • Michael Walker
    • Michael Walker
    • G10L1520
    • G10L25/48G10L2025/783
    • Speech level measurement is particularly significant for successful echo compensation in telecommunications systems, for noise suppression in a noisy environment, for example in military vehicles, or in speech recognition and in speech coding and decoding systems. A method is indicated which permits speech levels measurement only if features of speech are recognized and interferences and speech pauses are filtered out for the measurement. To this end, speech and pause detectors and a mean value generator are utilized, the time behavior of which is largely adapted to the perception capability of the human ear. Briefly spoken vowels thus are well detected, while nasal sounds or consonants are suppressed in the case of falling levels. A speech level measuring device is indicated which provides very accurate results in a short adaptation period.
    • 语音电平测量对于电信系统中成功的回波补偿,例如在嘈杂环境中的噪声抑制,例如在军用车辆中,或在语音识别和语音编码和解码系统中尤其重要。 指示一种方法,其仅在语音特征被识别并且干扰和语音暂停被滤出用于测量时才允许语音水平测量。 为此,利用语音和暂停检测器和平均值发生器,其时间行为很大程度上适应于人耳的感知能力。 简单说出的元音因此被很好地发现,而鼻音或辅音在下降的情况下被压制。 指示语音电平测量装置,其在短的适应期内提供非常精确的结果。
    • 29. 发明授权
    • Dispensing apparatus for viscous liquids
    • 粘性液体分配装置
    • US06499629B1
    • 2002-12-31
    • US09578366
    • 2000-05-25
    • Paul K. ColangeloPeter J. PetreccaAlan RamspeckMichael Walker
    • Paul K. ColangeloPeter J. PetreccaAlan RamspeckMichael Walker
    • B67D562
    • B05C11/1034B05C5/001B05C5/02B05C5/0225Y10T137/2196Y10T137/6525Y10T137/6606
    • Apparatus for dispensing viscous liquid, such as hot melt adhesive, includes a manifold, a dispensing module, a heater thermally coupled to the manifold, and thermally insulating cover structure secured around both the module and the manifold. Air gaps are formed between the cover structure and the heated components inside to further reduce heat transfer. The cover structure may also include heat dissipating fins. A supply connector associated with the manifold includes an interior flow passage, an exterior annular recess and at least one port communicating therebetween. A valve includes a valve seat having an orifice and a sealing surface located around the orifice. The valve further includes a valve stem movable between open and closed positions and having a recess in one end and a sealing edge located around the recess. A valve module includes an integrated heating element for providing localized heat to the adhesive immediately prior to dispensing.
    • 用于分配粘性液体(例如热熔粘合剂)的设备包括歧管,分配模块,热联接到歧管的加热器以及固定在模块和歧管两端的隔热盖结构。 在盖结构和内部加热部件之间形成气隙,以进一步减少热传递。 盖结构还可以包括散热翅片。 与歧管相关联的供应连接器包括内部流动通道,外部环形凹部和在其间连通的至少一个端口。 阀包括具有孔的阀座和位于孔周围的密封表面。 阀还包括可在打开位置和关闭位置之间移动的阀杆,并且在一端具有凹部,以及位于凹部周围的密封边缘。 阀模块包括集成的加热元件,用于在分配之前立即向粘合剂提供局部热量。
    • 30. 发明授权
    • Method of providing adaptive echo cancellation
    • 提供自适应回波消除的方法
    • US5570423A
    • 1996-10-29
    • US519303
    • 1995-08-25
    • Michael WalkerHans-Jurgen Matt
    • Michael WalkerHans-Jurgen Matt
    • H04M9/08H04B3/20
    • H04M9/082
    • A method of adaptive echo cancellation finds application for example in the hands-free facilities of telecommunications terminals, to compensate for disturbing echoes from the acoustic feedback between loudspeaker (1) and microphone (2). It is known to simulate these echoes (h(k)) with digital filters, and to subtract them from the echo-affected microphone signal (z(k)). The realization usually requires a signal processor with a high operating speed, a floating point logic and a large memory space. The task is to present a cost-effective method, which can be used under different acoustic conditions. The invention fulfills this task by using a known FIR filter (5), whose filter coefficients (c.sub.1 to c.sub.N) are determined according to the NLMS algorithm. The step width (.alpha.), which determines the changing speed of the filter coefficients (c.sub.1 to c.sub.N), can be made a function of one or more of various factors, including the current loudspeaker-microphone coupling (dlm), a quality criterion (Q) of the excitation, the existence of local speech signals (lokspk), the existence of briefly fluctuating local noises (lnoise), and the efficiency of the echo canceller. A special scaling of the coefficient values in conjunction with the adaptive step width control, advantageously permits the use of short FIR filters and cost-effective fixed-point computers.
    • 自适应回波消除的方法例如在电信终端的免提设施中应用,以补偿来自扬声器(1)和麦克风(2)之间的声反馈的干扰回波。 已知用数字滤波器模拟这些回波(h(k)),并从回波影响的麦克风信号(z(k))中减去它们。 该实现通常需要具有高操作速度,浮点逻辑和大的存储空间的信号处理器。 任务是提供一种经济有效的方法,可以在不同的声学条件下使用。 本发明通过使用已知的FIR滤波器(5)来实现该任务,其滤波器系数(c1至cN)根据NLMS算法确定。 确定滤波器系数(c1至cN)的变化速度的步长(α)可以作为各种因素中的一种或多种的函数,包括当前的扬声器 - 麦克风耦合(dlm),质量标准(Q ),本地语音信号(lokspk)的存在,短暂波动的局部噪声(lnoise)的存在以及回声消除器的效率。 结合自适应步长控制的系数值的特殊缩放有利地允许使用短FIR滤波器和成本有效的定点计算机。