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    • 2. 发明授权
    • Subband acoustic feedback cancellation in hearing aids
    • 助听器中的子带声反馈消除
    • US06480610B1
    • 2002-11-12
    • US09399483
    • 1999-09-21
    • Xiaoling FangGerald WilsonBrad Giles
    • Xiaoling FangGerald WilsonBrad Giles
    • H04R2500
    • H04R25/453H04R25/505H04R2430/03
    • A new subband feedback cancellation scheme is proposed, capable of providing additional stable gain without introducing audible artifacts. The subband feedback cancellation scheme employs a cascade of two narrow-band filters Ai(Z) and Bi(Z) along with a fixed delay, instead of a single filter Wi(Z) and a delay to represent the feedback path in each subband. The first filter, Ai(Z), is called the training filter, and models the static portion of the feedback path in ith subband, including microphone, receiver, ear canal resonance, and other relatively static parameters. The training filter can be implemented as a FIR filter or as an IIR filter. The second filter, BI(Z), is called a tracking filter and is typically implemented as a FIR filter with fewer taps than the training filter. This second filter tracks the variations of the feedback path in the ith subband caused by jaw movement or objects close to the ears of the user.
    • 提出了一种新的子带反馈消除方案,能够提供额外的稳定增益而不引入可听见的伪像。 子带反馈消除方案采用两个窄带滤波器Ai(Z)和Bi(Z)的级联以及固定延迟,而不是单个滤波器Wi(Z)以及用于表示每个子带中的反馈路径的延迟。 第一个滤波器Ai(Z)被称为训练滤波器,并对第i个子带中的反馈路径的静态部分进行建模,包括麦克风,接收机,耳道共振和其他相对静态的参数。 训练滤波器可以实现为FIR滤波器或IIR滤波器。 第二个滤波器BI(Z)被称为跟踪滤波器,通常被实现为具有比训练滤波器少的抽头的FIR滤波器。 该第二滤波器跟踪由钳口移动引起的第i子带中的反馈路径的变化或靠近用户耳朵的物体。
    • 3. 发明授权
    • Multiplierless interpolator for a delta-sigma digital to analog converter
    • 用于delta-sigma数模转换器的无量纲内插器
    • US06392576B1
    • 2002-05-21
    • US09935095
    • 2001-08-21
    • Gerald WilsonRobert S. Green
    • Gerald WilsonRobert S. Green
    • H03M300
    • H03H17/0225H03H17/0279H03H17/0285H03H17/0416H03H17/0444
    • A simplified algorithm for digital signal interpolation and a novel architecture to implement the algorithm in an integrated circuit (“IC”) with significant space constraints are presented. According to embodiments of the present invention, the interpolator is divided into two parts. The first part of the interpolator increases the sample rate by a factor of two and smoothes the signal using a half-band Infinite Impulse Response (“IIR”) filter. The second part of the interpolator increases the sample rate of the signal by a factor of thirty-two using a zero-order-hold (“ZOH”) circuit. In one embodiment, the half-band IIR filter is implemented using an all-pass lattice structure to minimize quantization effects. The lattice coefficients are chosen such that the structure can achieve all filter design requirements, yet is capable of being implemented with a small number of shifters and adders, and no multipliers.
    • 提出了一种用于数字信号插值的简化算法和一种在具有显着空间约束的集成电路(“IC”)中实现该算法的新型架构。 根据本发明的实施例,内插器被分成两部分。 插值器的第一部分将采样率提高一倍,并使用半带无限脉冲响应(“IIR”)滤波器对信号进行平滑。 内插器的第二部分使用零级保持(“ZOH”)电路将信号的采样率增加了三十二倍。 在一个实施例中,使用全通格格结构来实现半带IIR滤波器以使量化效应最小化。 选择晶格系数使得该结构可以实现所有滤波器设计要求,但是能够用少量的移位器和加法器实现,并且不需要乘法器。
    • 6. 发明申请
    • HEARING AID NOISE REDUCTION METHOD, SYSTEM, AND APPARATUS
    • 听力噪声减少方法,系统和装置
    • US20090220114A1
    • 2009-09-03
    • US12040507
    • 2008-02-29
    • Gerald Wilson
    • Gerald Wilson
    • H04R25/00
    • H04R25/453
    • A computer-implemented method including receiving a first signal from an input device of a hearing aid. The first signal may include a noise signal. The computer-implemented method may include low-pass filtering first periodic samples of the first signal, and the first periodic samples may be approximately periodic with respect to a period of the noise signal. The computer-implemented method may further include low-pass filtering second periodic samples of the first signal, and the second periodic samples may be approximately periodic with respect to the period of the noise signal. The second periodic samples may also be phase shifted relative to the first periodic samples. Hearing aid systems and apparatuses are also disclosed.
    • 一种计算机实现的方法,包括从助听器的输入装置接收第一信号。 第一信号可以包括噪声信号。 计算机实现的方法可以包括第一信号的低通滤波第一周期性样本,并且第一周期性样本可以相对于噪声信号的周期近似周期性。 计算机实现的方法还可以包括第一信号的低通滤波第二周期性样本,并且第二周期性样本可以相对于噪声信号的周期近似周期性。 第二周期性样本也可以相对于第一周期性样本相移。 还公开了助听器系统和装置。
    • 8. 发明授权
    • Multiplierless interpolator for a delta-sigma digital to analog converter
    • 用于delta-sigma数模转换器的无量纲内插器
    • US06313773B1
    • 2001-11-06
    • US09491695
    • 2000-01-26
    • Gerald WilsonRobert S. Green
    • Gerald WilsonRobert S. Green
    • H03M300
    • H03H17/0225H03H17/0279H03H17/0285H03H17/0416H03H17/0444
    • A simplified algorithm for digital signal interpolation and a novel architecture to implement the algorithm in an integrated circuit (“IC”) with significant space constraints are presented. According to embodiments of the present invention, the interpolator is divided into two parts. The first part of the interpolator increases the sample rate by a factor of two and smoothes the signal using a half-band Infinite Impulse Response (“IIR”) filter. The second part of the interpolator increases the sample rate of the signal by a factor of thirty-two using a zero-order-hold (“ZOH”) circuit. In one embodiment, the half-band IIR filter is implemented using an all-pass lattice structure to minimize quantization effects. The lattice coefficients are chosen such that the structure can achieve all filter design requirements, yet is capable of being implemented with a small number of shifters and adders, and no multipliers.
    • 提出了一种用于数字信号插值的简化算法和一种在具有显着空间约束的集成电路(“IC”)中实现该算法的新型架构。 根据本发明的实施例,内插器被分成两部分。 插值器的第一部分将采样率提高一倍,并使用半带无限脉冲响应(“IIR”)滤波器对信号进行平滑。 内插器的第二部分使用零级保持(“ZOH”)电路将信号的采样率增加了三十二倍。 在一个实施例中,使用全通格格结构来实现半带IIR滤波器以使量化效应最小化。 选择晶格系数使得该结构可以实现所有滤波器设计要求,但是能够用少量的移位器和加法器实现,并且不需要乘法器。
    • 9. 发明授权
    • Hearing aid noise reduction method, system, and apparatus
    • 助听器降噪方法,系统和装置
    • US08340333B2
    • 2012-12-25
    • US12040507
    • 2008-02-29
    • Gerald Wilson
    • Gerald Wilson
    • H04R25/00
    • H04R25/453
    • A computer-implemented method including receiving a first signal from an input device of a hearing aid. The first signal may include a noise signal. The computer-implemented method may include low-pass filtering first periodic samples of the first signal, and the first periodic samples may be approximately periodic with respect to a period of the noise signal. The computer-implemented method may further include low-pass filtering second periodic samples of the first signal, and the second periodic samples may be approximately periodic with respect to the period of the noise signal. The second periodic samples may also be phase shifted relative to the first periodic samples. Hearing aid systems and apparatuses are also disclosed.
    • 一种计算机实现的方法,包括从助听器的输入装置接收第一信号。 第一信号可以包括噪声信号。 计算机实现的方法可以包括第一信号的低通滤波第一周期性样本,并且第一周期性样本可以相对于噪声信号的周期近似周期性。 计算机实现的方法还可以包括第一信号的低通滤波第二周期性样本,并且第二周期性样本可以相对于噪声信号的周期近似周期性。 第二周期性样本也可以相对于第一周期性样本相移。 还公开了助听器系统和装置。
    • 10. 发明授权
    • Subband acoustic feedback cancellation in hearing aids
    • 助听器中的子带声反馈消除
    • US07020297B2
    • 2006-03-28
    • US10737206
    • 2003-12-15
    • Xiaoling FangGerald WilsonBrad Giles
    • Xiaoling FangGerald WilsonBrad Giles
    • H04R25/00
    • H04R25/453H04R25/505H04R2430/03
    • A new subband feedback cancellation scheme is proposed, capable of providing additional stable gain without introducing audible artifacts. The subband feedback cancellation scheme employs a cascade of two narrow-band filters Ai(Z) and Bi(Z) along with a fixed delay, instead of a single filter Wi(Z) and a delay to represent the feedback path in each subband. The first filter, Ai(Z), is called the training filter, and models the static portion of the feedback path in ith subband, including microphone, receiver, ear canal resonance, and other relatively static parameters. The training filter can be implemented as a FIR filter or as an IIR filter. The second filter, BI(Z), is called a tracking filter and is typically implemented as a FIR filter with fewer taps than the training filter. This second filter tracks the variations of the feedback path in the ith subband caused by jaw movement or objects close to the ears of the user.
    • 提出了一种新的子带反馈消除方案,能够提供额外的稳定增益而不引入可听见的伪像。 子带反馈消除方案采用两个窄带滤波器A SUB(Z)和B SUB(Z)的级联以及固定延迟,而不是单个滤波器 (Z)和延迟以表示每个子带中的反馈路径。 第一个滤波器A(N)被称为训练滤波器,并对第i个子频带中的反馈路径的静态部分进行建模,包括麦克风,接收机,耳机 运河共振等相对静态的参数。 训练滤波器可以实现为FIR滤波器或IIR滤波器。 第二个滤波器B 1(Z)被称为跟踪滤波器,并且通常被实现为具有比训练滤波器更少的抽头的FIR滤波器。 该第二滤波器跟踪由钳口移动引起的第i / SUP子频带中的反馈路径的变化或靠近用户耳朵的物体。