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    • 21. 发明申请
    • CHANNEL ESTIMATION ENHANCED LMS EQUALIZER
    • 信道估计增强LMS均衡器
    • WO2006101997A3
    • 2008-01-24
    • PCT/US2006009556
    • 2006-03-16
    • INTERDIGITAL TECH CORPPIETRASKI PHILIP J
    • PIETRASKI PHILIP J
    • H04B1/10H03H7/30
    • H03H21/0043H03H2021/0056H04L25/0226H04L25/03019H04L2025/03496H04L2025/03611
    • The present invention is related to an enhanced equalizer (106) using channel estimation. A scaled version of a channel estimate is used as an expected average behavior of the product of a transmitted signal and a received signal to implement Griffith algorithm. The present invention also uses advance or prediction of a channel estimate (112) to overcome the lag problem inherent in a least means square (LMS) algorithm in a time varying channel. Therefore, the present invention enables the use of a small step size while attaining the same tracking capability with a large step size. A channel estimate at some time in the future is used for updating equalizer filter tap coefficients. This may be performed with a prediction filter. Alternatively, a delay (104) may be introduced in the input data to the filter tap coefficient generator (114), which makes a channel estimate look like a prediction to the filter tap coefficient generator.
    • 本发明涉及使用信道估计的增强均衡器(106)。 信道估计的缩放版本被用作发射信号和接收信号的乘积的预期平均行为以实现Griffith算法。 本发明还使用信道估计(112)的提前或预测来克服时变信道中最小均方(LMS)算法中固有的滞后问题。 因此,本发明能够在以大的步长获得相同的跟踪能力的同时使用小的步长。 将来某个时间的信道估计用于更新均衡器滤波器抽头系数。 这可以用预测滤波器来执行。 或者,可以在输入数据中将延迟(104)引入到滤波器抽头系数产生器(114),这使得信道估计看起来像对滤波器抽头系数发生器的预测。
    • 22. 发明申请
    • システム推定方法及びプログラム及び記録媒体、システム推定装置
    • 系统估计方法,程序,记录介质,系统估计装置
    • WO2005015737A1
    • 2005-02-17
    • PCT/JP2004/011568
    • 2004-08-05
    • 独立行政法人科学技術振興機構西山 清
    • 西山 清
    • H03H21/00
    • H04B3/23G05B13/024H03H21/0043H03H2021/005Y10S367/901
    • 忘却係数を理論的に最適に決定できる推定方法を確立すると共に、その数値的に安定な推定アルゴリズムと高速アルゴリズムを開発する。まず、処理部は、記憶部又は入力部ら上限値γ f を読み出し又は入力する(S101)。処理部は、式(15)によって忘却係数ρを決定する(S103)。その後、処理部は、忘却係数ρに基づき、式(10)~式(13)のハイパーH ∞ フィルタを実行する(S105)。処理部101は、式(17)(あるいは、後述の式(18))の存在条件を計算し(S107)、その存在条件がすべての時刻で満たされれば(S109)、γ f をΔγだけ小さくして同じ処理を繰り返す(S111)。一方、あるγ f で存在条件を満たさなくなったとき(S109)、そのγ f にΔγを加えたものをγ f の最適値γ f op として出力部に出力及び/又は記憶部に記憶する(S113)。
    • 可以建立一种能够逻辑和最佳地决定遗忘系数并开发数值稳定的估计算法和高速算法的估计方法。 首先,处理部从存储部或输入部读出或接收上限值gammaf(S101)。 处理部通过式(15)决定遗忘系数rho(S103)。 之后,根据遗忘系数rho,处理部执行等式(10-13)的超H∞滤波器(S105)。 处理部(101)计算等式(17)(或稍后给出的等式(18))的存在条件(S107)。 当存在条件始终满足时(S109),通过Deltagamma减少gammaf并重复相同的处理(S111)。 另一方面,当存在条件不满足某个gammaf(S109)时,将Deltagamma添加到gammaf中,并且将和输出到输出部分和/或存储在存储部分中作为最佳值gammaf < (S113)。
    • 23. 发明申请
    • SYSTEM IDENTIFYING METHOD
    • 系统识别方法
    • WO02035727A1
    • 2002-05-02
    • PCT/JP2001/009082
    • 2001-10-16
    • G05B13/02H03H21/00H04B3/23
    • H03H21/0043H04B3/23
    • Quick real-time identification and estimation of a time-non-varying or time-varying system. A new H INFINITY evaluation criterion is determined, a fast algorithm for a modified H INFINITY filter based on the criterion is developed, and a quick time-varying system identifying method according to the fast H INFINITY filtering algorithm is provided. By the fast H INFINITY filtering algorithm, a time-varying system sharply varying can be traced with an amount of calculation O(N) per unit time step. The algorithm completely agrees with a fast Kalman filtering algorithm at the extreme of the upper limit value. If the estimate of impulse response is determined, a pseudo-echo is sequentially determined from the estimate and subtracted from the real echo to cancel the echo. Thus an echo chancellor is realized.
    • 快速实时识别和估计时变不变或时变系统。 确定了一个新的H INFINITY评估标准,开发了基于该标准的修改后的H INFINITY滤波器的快速算法,并提供了一种基于快速H INFINITY滤波算法的快速时变系统识别方法。 通过快速H INFINITY滤波算法,可以用每单位时间步长的计算量O(N)追踪急剧变化的时变系统。 该算法完全符合在极限值上的快速卡尔曼滤波算法。 如果确定了脉冲响应的估计,则从估计中依次确定伪回波并从真实回波中减去以消除回波。 因此,实现了回音总理。
    • 24. 发明申请
    • ADAPTIVE FILTERING METHOD BASED ON SIGN ALGORITM AND RELATED DEVICE
    • 基于标识符和相关设备的自适应滤波方法
    • WO0184708A3
    • 2002-03-21
    • PCT/EP0104370
    • 2001-04-17
    • KONINKL PHILIPS ELECTRONICS NV
    • GORNSTEIN VIKTOR LTURKENICH GENNADY
    • H03H21/00
    • H03H21/0043
    • An input signal is filtered for creating an output signal using an adaptive filter. An error signal is derived from the output signal. The adaptive filter has coefficient whose value can be modified. A value of a coefficient is modified using a derived updating amount. The updating amount is obtained from the product of a value of the input signal, a value of the polarity of the error signal, and a step gain. The step gain has the form 2 with K being an integer and being dependent on a magnitude of the value of the error signal and on a step gain parameter. The updating amount is dependent on both the magnitude and the polarity of the error signal, therefore allowing a precise update of the coefficient. The specific form of the step gain allows a fast derivation of the product.
    • 对输入信号进行滤波,以使用自适应滤波器产生输出信号。 从输出信号导出误差信号。 自适应滤波器的值可以被修改。 使用导出的更新量来修改系数的值。 从输入信号的值,误差信号的极性的值和阶跃增益的乘积得到更新量。 阶梯增益具有形式2 K,其中K是整数,并且取决于误差信号的值的大小和阶跃增益参数。 更新量取决于误差信号的幅度和极性,从而允许精确地更新系数。 阶梯增益的具体形式允许快速推导产品。
    • 27. 发明申请
    • FILTER COEFFICIENT UPDATING IN TIME DOMAIN FILTERING
    • 过滤系统在时域过滤中的更新
    • WO2017048997A1
    • 2017-03-23
    • PCT/US2016/051994
    • 2016-09-15
    • DOLBY LABORATORIES LICENSING CORPORATION
    • SHI, DongSUN, Xuejing
    • G10L25/48G10L25/18
    • G10L25/48G10L25/18H03H17/0294H03H21/0043
    • Example embodiments disclosed herein relate to filter coefficient updating in time domain filtering. A method of processing an audio signal is disclosed. The method includes obtaining a predetermined number of target gains for a first portion of the audio signal by analyzing the first portion of the audio signal. Each of the target gains is corresponding to a linear subband of the audio signal. The method also includes determining a filter coefficients for time domain filtering the first portion of the audio signal so as to approximate a frequency response given by the target gains. The filter coefficients are determined by iteratively selecting at least one target gain from the target gains and updating the filter coefficient based on the selected at least one target gain. Corresponding system and computer program product for processing an audio signal are also disclosed.
    • 本文公开的示例实施例涉及时域滤波中的滤波器系数更新。 公开了一种处理音频信号的方法。 该方法包括通过分析音频信号的第一部分来获得音频信号的第一部分的预定数目的目标增益。 每个目标增益对应于音频信号的线性子带。 该方法还包括确定用于对音频信号的第一部分进行时域滤波的滤波器系数,以便近似由目标增益给出的频率响应。 通过从目标增益迭代地选择至少一个目标增益并基于所选择的至少一个目标增益来更新滤波器系数来确定滤波器系数。 还公开了用于处理音频信号的相应系统和计算机程序产品。
    • 29. 发明申请
    • ADAPTIVE SYSTEMS USING CORRENTROPY
    • 使用CORRENTROPY的自适应系统
    • WO2011100491A2
    • 2011-08-18
    • PCT/US2011024435
    • 2011-02-11
    • UNIV FLORIDAPRINCIPE JOSE CARLOSSINGH ABHISHEKLIU WEIFENG
    • PRINCIPE JOSE CARLOSSINGH ABHISHEKLIU WEIFENG
    • H03H21/00
    • G10L21/0264H03H21/0043
    • Various methods and systems are provided for related to adaptive systems using correntropy. In one embodiment, a signal processing device includes a processing unit and a memory storing an adaptive system executable in the at least one processing unit. The adaptive system includes modules that, when executed by the processing unit, cause the signal processing device to adaptively filter a desired signal using a correntropy cost function. In another embodiment, a method includes adjusting a coefficient of an adaptive filter based at least in part on a correntropy cost function signal, providing an adaptive filter output signal based at least in part on the adjusted coefficient and a reference signal, and determining an error signal based at least in part on a received signal and the adaptive filter output signal.
    • 提供了各种方法和系统,用于与使用correntropy的自适应系统相关。 在一个实施例中,信号处理设备包括处理单元和存储可在该至少一个处理单元中执行的自适应系统的存储器。 自适应系统包括当由处理单元执行时使得信号处理装置使用科学成本函数自适应地过滤所需信号的模块。 在另一个实施例中,一种方法包括至少部分地基于科学成本函数信号来调整自适应滤波器的系数,至少部分地基于经调整的系数和参考信号提供自适应滤波器输出信号,并且确定误差 信号至少部分地基于接收信号和自适应滤波器输出信号。