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    • 21. 发明授权
    • Non-linear deterministic stochastic filtering method and system
    • 非线性确定性随机过滤方法和系统
    • US5721694A
    • 1998-02-24
    • US241179
    • 1994-05-10
    • Daniel Graupe
    • Daniel Graupe
    • G10L21/02H03H21/00H04L25/03H04B15/00
    • G10L21/0208H03H21/0016H03H21/0027H04L25/03159H04L25/03993H04L2025/03522
    • A method and system for filtering a noisy input comprised of an information signal and a noise signal. No prior time domain value information or frequency domain parameter information on the information signal or noise signal is available. A whitening filter renders the noise part of a noise signal into a substantially white noise signal. A fourier transform system provides an amplitude spectrum output and a phase output. A multiplier squares the amplitude spectrum output thus providing a power spectrum output. A zero mean sub-system generates a zero mean power spectrum output from the power spectrum output. A square root sub-system squares the zero mean power spectrum output thus providing a resultant output. An inverse Fourier transform sub-system inversely transforms the resultant output to provide a time domain resultant output. An inverse whitening filter provides a substantially noise free output from the time domain resultant output.
    • 一种滤波由信息信号和噪声信号构成的噪声输入的方法和系统。 没有关于信息信号或噪声信号的先前时域信息或频域参数信息可用。 白化滤波器将噪声信号的噪声部分呈现为基本上白噪声的信号。 傅立叶变换系统提供幅度谱输出和相位输出。 乘法器对幅度谱输出进行平方,从而提供功率谱输出。 零均值子系统从功率谱输出产生零平均功率谱输出。 平方根子系统对零平均功率谱输出进行平方,从而提供合成输出。 逆傅里叶变换子系统对所得到的输出进行反变换以提供时域结果输出。 反向白化滤波器从时域结果输出提供基本上无噪声的输出。
    • 22. 发明授权
    • Microprocessor-controlled multiplexed functional electrical stimulator
for surface stimulation in paralyzed patients
    • US5014705A
    • 1991-05-14
    • US335197
    • 1989-04-07
    • Daniel GraupeAmihadar Arber
    • Daniel GraupeAmihadar Arber
    • A61N1/00A61F4/00A61N1/36
    • A61F4/00A61N1/36003
    • In accordance with the present invention, a functional electrical stimulation system enables upper-motor-neuron paraplegic patients and certain other paralyzed or partially paralyzed patients to stand and to take steps with the support of a walker in an independent fashion so that they are in control of the stimulation and, noting that they have very little or not sensation, that they still can obtain some feedback on the state of the system. In one embodiment, a compact and safe battery-operated stimulation system is provided that is of pocket size and that allows for non-invasive safe and controlled stimulation using skin surface electrodes.The system of the present invention is a stimulation system for providing upper-motor-neuron paralyzed patients with capabilities of unbraced standing and unbraced walking, which consists of a single pulse power and amplifier that outputs various trains of stimulation pulses and which is totally under microprocessor generation and control, where the microprocessor's microcomputer selectively performs multiplexing of the pulse trains and their selective alternate distribution to the various surface electrodes attached to the skin of the patient at the appropriate stimulation locations. It also controls pulse-width, pulse-duration, pulse-amplitude and pulse-shape.Furthermore, the system can compute and activate (audible and/or visual) warning signals to warn the patient when the system is close to its maximal output. This would happen when muscle fatigue is such that no further increase in stimuli level can keep the patient upright. The system computes and outputs indications of the various stimuli levels to a visual display, and it provides automatic fail-safe and override capabilities to help ensure that the patient does not fall in case of major circuit or system failure or misactivation. This is a critical capability since this sytem will be used by paraplegics who lack sensation in their lower extremities.
    • 23. 发明授权
    • Multi-function control system for an artificial upper-extremity
prosthesis for above-elbow amputees
    • 用于上肘截肢者人造上肢假体的多功能控制系统
    • US4030141A
    • 1977-06-21
    • US656412
    • 1976-02-09
    • Daniel Graupe
    • Daniel Graupe
    • A61F2/68A61F2/70A61F2/72A61F1/00A61F1/06
    • A61F2/68A61F2/72A61F2002/704
    • This invention relates to an improved EMG-based method for actuating a prosthetic appliance wherein a single pair of electrodes may be used to actuate several artificial limb movements by way of sampling and repeatedly resampling substantially the entire time function of the myoelectric signal as it appears on the limb stump over a period so short that the signal remains quasi-stationary, comparing a minimum number of parameters of such signals with a range of values of these same parameters stored within the data bank of a microcomputer, and actuating one or more functions of a prosthetic appliance whenever the current data for the sampled parameters all falls within a preselected domain chosen on the basis of the historical data. This invention also encompasses the apparatus for using the EMG signals developed by the patient in the stump for controlling the movements of a prosthetic appliance which comprise means connected to process the EMG signals so as to differentiate between the various functions to be performed thereby based upon a minimum number of parameters of said signals, means for storing historical data concerning the range of values said parameters vary over for each function, means for repeatedly sampling current data on said parameters and comparing same with the stored data during a brief time interval, and means operative to initiate one or more functions whenever the current data on the chosen parameters falls within an arbitrary domain chosen on the basis of said historical data.
    • 本发明涉及一种用于致动假体器具的改进的基于EMG的方法,其中可以使用单对电极来驱动几个假肢运动,通过取样并基本上重复地重新采样肌电信号的整个时间函数,因为它出现在 肢体残端在一段短暂的时间内使得信号保持准静止,将这种信号的参数的最小数量与存储在微型计算机的数据库内的这些相同参数的值的范围进行比较,并且执行一个或多个功能 每当采样参数的当前数据都落在基于历史数据选择的预选域内时,假肢器具。 本发明还包括用于使用患者在残端中开发的EMG信号的装置,用于控制假肢器具的运动,该装置包括连接到处理EMG信号的装置,以便基于以下步骤来区分要执行的各种功能: 所述信号的参数的最小数量,用于存储关于所述参数的范围的历史数据的装置,用于每个功能的所述参数变化,用于重复地对所述参数上的当前数据进行采样并在短时间间隔内与存储的数据进行比较的装置, 每当所选择的参数上的当前数据落在基于所述历史数据选择的任意域内时,起作用地启动一个或多个功能。
    • 25. 发明授权
    • Apparatus for managing a neurological disorder
    • 用于管理神经障碍的装置
    • US08391986B2
    • 2013-03-05
    • US12992265
    • 2009-05-12
    • Daniel GraupeDaniela TuninettiIshita Basu
    • Daniel GraupeDaniela TuninettiIshita Basu
    • A61N1/00
    • A61B5/0478A61B5/0492A61B5/1101A61B5/4082A61B5/726A61N1/36025A61N1/36082
    • A system that incorporates teachings of the present disclosure may include, for example, a Neuro Sensing and Stimulation Device having a conductive lead for implanting in a brain of a living entity, a sensor coupled to the conductive lead for collecting data associated with one or more electrical signals detected in the conductive lead, a stimulator coupled to the conductive lead to apply one or more electrical signals thereto, and a controller. The controller can be operable to collect data associated with one or more electrical signals detected in the conductive lead, determine a stimulation sequence from the collected data, and direct the stimulator to apply on the conductive lead one or more electrical signals determined from the stimulation sequence. Additional embodiments are disclosed.
    • 结合本公开的教导的系统可以包括例如具有用于植入活体的脑中的导电引线的神经感觉和刺激装置,耦合到导电引线的传感器,用于收集与一个或多个相关联的数据 在导电引线中检测到的电信号,耦合到导电引线的刺激器向其施加一个或多个电信号,以及控制器。 所述控制器可操作以收集与在所述导电引线中检测到的一个或多个电信号相关联的数据,从所收集的数据确定刺激序列,并且将所述刺激器引导到所述导电引线上施加一个或多个从所述刺激序列确定的电信号 。 公开了另外的实施例。
    • 27. 发明授权
    • Compression/decompression for preservation of high fidelity speech
quality at low bandwidth
    • 压缩/解压缩,以保持低带宽的高保真语音质量
    • US5822370A
    • 1998-10-13
    • US632914
    • 1996-04-16
    • Daniel Graupe
    • Daniel Graupe
    • G10L19/02H04B1/66
    • G10L19/0204G10L25/27G10L25/51
    • The input signal is filtered by bandpass filters having different passbands. These filtered signals are input to power detectors that average the power present in each band. A comparator compares each power level signal to a predetermined power threshold to determine if information is present in any of the bands. If information is present in the upper bands, the information is transformed by a discrete wavelet transform and is thresholded and then shifted to the lower bands. The process by which the shifting operation was accomplished is stored in a code book band. An inverse wavelet transform generates the compressed signal by transforming the signals from the wavelet domain to the time domain. If the signal was compressed, the code book signal is transmitted with the compressed signal to a receiving unit for decompression. If the signal was not compressed, the code book signal and the original input signal is transmitted to the receiving unit. The receiving unit receives the transmitted signal and reconstructs the original input from the transmitted signal, either directly or by re-spreading and transforming the compressed signal from the transmitted signal responsive to the code book signal embedded with the transmitted signal.
    • 输入信号由具有不同通带的带通滤波器滤波。 这些滤波的信号被输入到功率检测器,其平均每个频带中存在的功率。 比较器将每个功率电平信号与预定功率阈值进行比较,以确定信息是否存在于任何频带中。 如果信息存在于较高频带中,则信息通过离散小波变换进行变换,并被阈值化,然后转移到较低频带。 完成换档操作的过程被存储在码本频带中。 逆小波变换通过将来自小波域的信号变换为时域来产生压缩信号。 如果信号被压缩,则将码本信号与压缩信号一起发送到用于解压缩的接收单元。 如果信号未被压缩,则码本信号和原始输入信号被发送到接收单元。 接收单元接收所发送的信号,并且根据所发送的信号嵌入的码本信号,直接地或通过重新扩展和变换来自所发送的信号的压缩信号,从发送的信号中重构原始输入。
    • 28. 发明授权
    • Artificial intelligence pattern-recognition-based noise reduction system
for speech processing
    • 基于人工智能模式识别的语音处理降噪系统
    • US5097510A
    • 1992-03-17
    • US432525
    • 1989-11-07
    • Daniel Graupe
    • Daniel Graupe
    • G10L21/02
    • G10L21/0208G10L21/0232
    • A system is provided to reduce noise from a signal of speech that is contaminated by noise. The present system employs an artificial intelligence that is capable of deciding upon the adjustment of a filter subsystem by distinguishing between noise and speech in the spectrum of the incoming signal of speech plus noise. The system does this by testing the pattern of a power or envelope function of the frequency spectrum of the incoming signal. The system determines that the fast changing portions of that envelope denote speech whereas the residual is determined to be the frequency distribution of the noise power. This determination is done while examining either the whole spectrum, or frequency bands thereof, regardless of where the maximum of the spectrum lies. In another embodiment of the invention, a feedback loop is incorporated which provides incremental adjustments to the filter by employing a gradient search procedure to attempt to increase certain speech-like features in the system's output. The present system does not require consideration of minima of functions of the incoming signal or pauses in speech. Instead, the present system employs an artificial intelligence system to which is input the envelope pattern of the incoming signal of speech and noise. The present system then filters out of this envelope signal the rapidly changing variations of the envelope over fixed time windows.
    • 提供一种系统来减少被噪声污染的语音信号的噪声。 本系统采用人造智能,其能够通过区分语音加噪声的输入信号的频谱中的噪声和语音来决定滤波器子系统的调整。 系统通过测试输入信号的频谱的功率或包络函数的模式来实现。 系统确定该包络的快速变化部分表示语音,而残差被确定为噪声功率的频率分布。 无论频谱的最大值位于何处,都可以检查整个频谱或其频带。 在本发明的另一个实施例中,结合了反馈回路,其通过采用梯度搜索过程来尝试增加系统输出中的某些类似语音的特征来向滤波器提供增量调整。 本系统不需要考虑输入信号的功能的最小值或言语中的暂停。 相反,本系统采用人造智能系统,其输入语音和噪声的输入信号的包络图案。 然后,本系统从固定时间窗口中滤出包络信号中快速变化的变化。
    • 29. 发明授权
    • ARMA filter and method for designing the same
    • ARMA滤波器及其设计方法
    • US4188667A
    • 1980-02-12
    • US852917
    • 1977-11-18
    • Daniel GraupeAloysius A. BeexG. Donald Causey
    • Daniel GraupeAloysius A. BeexG. Donald Causey
    • H04R25/04H03H11/04H03H17/02H03H17/04H04R25/00G06F15/34
    • H04R25/505H03H17/0258H03H17/04H04R25/453
    • A near minimum order ARMA type recursive filter with guaranteed stability and convergence is provided together with a method for obtaining the parameters of such filter. The amplitude/frequency response of the filter approximates an arbitrarily selected frequency spectrum of amplitude, and the phase response approximates a substantially linear function of frequency with an arbitrarily selected slope because the parameters are identified, off-line, using a minimization process that minimizes an integral error norm. The first step involves performing an inverse discrete Fourier transform of the arbitrarily selected frequency spectrum of amplitude to obtain a truncated sequence of coefficients of a stable, pure moving-average filter model, i.e., the parameters of a non-recursive filter model. The truncated sequence of coefficients, which has N+1 terms, is then convolved with a random sequence to obtain an output sequence associated with the random sequence. A time-domain, convergent parameter identification is then performed, in a manner that minimizes an integral error function norm, to obtain the near minimum order parameters .alpha..sub.i and .beta..sub.j of the model having the desired amplitude- and phase-frequency responses, the parameters satisfying the relationship: ##EQU1## where: .alpha..sub.i is the ith auto-regressive parameter, and .beta..sub.j is the jth moving-average parameter, respectively, of an ARMA-type recursive filter; n and m denote the order of the auto-regressive and the moving-average parts of the ARMA model, respectively; y.sub.k and u.sub.k are associated elements of the kth element of the output sequence and the random sequence, respectively; k is an integer; and v is a shift integer selected to provide the desired slope of the phase response given by 2.pi.(v-N/2).
    • 提供具有保证稳定性和收敛性的近最小订单ARMA类型递归滤波器以及用于获得这种滤波器的参数的方法。 滤波器的幅度/频率响应近似于任意选择的幅度频谱,并且相位响应以任意选择的斜率近似于频率的基本上线性的函数,因为参数被离线地识别,使用最小化处理使得 积分误差范数。 第一步涉及执行任意选择的振幅频谱的离散傅里叶逆变换,以获得稳定的纯移动平均滤波器模型的截断序列,即非递归滤波器模型的参数。 具有N + 1项的截断的系数序列然后与随机序列进行卷积以获得与随机序列相关联的输出序列。 然后以最小化积分误差函数范数的方式执行时域收敛参数识别,以获得具有期望幅度和相位频率响应的模型的近似最小阶次参数αi和βj, 满足以下关系的参数:其中:αi是第i个自回归参数,βj分别是ARMA型递归滤波器的第j个移动平均参数; n和m分别表示ARMA模型的自回归和移动平均部分的顺序; yk和uk分别是输出序列的第k个元素和随机序列的相关元素; k是整数; 并且v是选择为提供由2π(v-N / 2)给出的相位响应的期望斜率的移位整数。
    • 30. 发明授权
    • Method and means for adaptively filtering near-stationary noise from an
information bearing signal
    • 用于从信息承载信号自适应过滤近平稳噪声的方法和装置
    • US4185168A
    • 1980-01-22
    • US866938
    • 1978-01-04
    • Daniel GraupeG. Donald Causey
    • Daniel GraupeG. Donald Causey
    • H03G5/16H04R25/00H04R27/00
    • H03G5/165H04R25/505H04R25/453
    • An input signal containing information such as speech or music as well as near-stationary noise is applied in parallel to a noise-analysis circuit and a noise-reduction circuit, each of which comprises a plurality of bandpass filters covering the range of frequencies associated with the information. The absolute value, or a function thereof, of the output of each bandpass filter in the noise-analysis circuit is produced and smoothed. The presence of near-stationary noise in the input signal is determined by examining the nature of the smoothed signal in each band assuming noise has a frequency spectrum which does not vary with time or varies only within a narrow range over a predetermined period of time with respect to the spectral parameters of the information signal. If noise is detected, the noise-analysis circuit identifies spectral parameters of the information and/or noise in each band using the smoothed signal therein.In the preferred embodiment of the invention, the bandpass filters of the noise-reduction circuit have gain elements that are adjusted in accordance with the identified parameters to minimize, under some continuous minimization criterion, the effect of the noise in the input signal thus enhancing intelligibility of the information therein. Minimization can be such that the gain-to-parameter relationships are similar to those in Weiner or Kalman filtering theory with a-priori knowledge of the noise, or of the noise and information, except that in this case, a-priori knowledge of the noise is acquired via identification and is not preassumed.
    • 包含诸如语音或音乐以及近似平稳噪声的信息的输入信号被并行地施加到噪声分析电路和降噪电路,每个噪声分析电路和噪声降低电路包括多个带通滤波器,覆盖与 信息。 产生并平滑了噪声分析电路中的每个带通滤波器的输出的绝对值或其功能。 通过检查每个频带中的平滑信号的性质来确定输入信号中的近平稳噪声的存在,假设噪声具有不随时间变化的频谱或仅在预定时间段内在窄范围内变化的频谱, 相对于信息信号的频谱参数。 如果检测到噪声,则噪声分析电路使用其中的平滑信号来识别每个频带中的信息和/或噪声的频谱参数。 在本发明的优选实施例中,降噪电路的带通滤波器具有根据所识别的参数进行调整的增益元件,以便在某些连续最小化标准下将输入信号中的噪声的影响最小化从而增强可懂度 的信息。 最小化可以使得增益参数关系类似于具有噪声或噪声和信息的先验知识的Weiner或卡尔曼滤波理论中的增益参数关系,除了在这种情况下,先验知识 噪音是通过识别获得的,不是预先使用的。