Modified volterra-wiener-hammerstein (MVWH) method for loudspeaker modeling and equalization
    1.
    发明申请
    Modified volterra-wiener-hammerstein (MVWH) method for loudspeaker modeling and equalization 失效
    用于扬声器建模和均衡的改进的volterra-wiener-hammerstein(MVWH)方法

    公开(公告)号:US20060274904A1

    公开(公告)日:2006-12-07

    申请号:US11446085

    申请日:2006-05-31

    申请人: Khosrow Lashkari

    发明人: Khosrow Lashkari

    IPC分类号: H04R29/00

    摘要: A method and apparatus for adaptive precompensation is disclosed. In one embodiment, the method comprises modifying operation of a predistortion filter in response to previous predistorted values and an original input signal, determining a precompensation error between the original input samples and the predicted loudspeaker output, and substantially reducing the precompensation error by using the exact inverse of a loudspeaker model that is a cascaded arrangement of at least one linear system with a non-linear system.

    摘要翻译: 公开了一种用于自适应预补偿的方法和装置。 在一个实施例中,该方法包括响应于先前的预失真值和原始输入信号修改预失真滤波器的操作,确定原始输入样本与预测的扬声器输出之间的预补偿误差,并且通过使用确切的 作为具有非线性系统的至少一个线性系统的级联布置的扬声器模型的倒数。

    Joint optimization of speech excitation and filter parameters
    2.
    发明授权
    Joint optimization of speech excitation and filter parameters 有权
    语音激励和滤波参数联合优化

    公开(公告)号:US07236928B2

    公开(公告)日:2007-06-26

    申请号:US10023826

    申请日:2001-12-19

    IPC分类号: G10L19/08

    CPC分类号: G10L19/10 G10L19/06

    摘要: An efficient optimization algorithm is provided for multipulse speech coding systems. The efficient algorithm performs computations using the contribution of the non-zero pulses of the excitation function and not the zeroes of the excitation function. Accordingly, efficiency improvements of 87% to 99% are possible with the efficient optimization algorithm.

    摘要翻译: 为多脉冲语音编码系统提供了一种有效的优化算法。 有效的算法使用激励函数的非零脉冲的贡献而不是激励函数的零来执行计算。 因此,有效的优化算法可以提高87%到99%的效率。

    Method and apparatus for frame-based loudspeaker equalization
    3.
    发明申请
    Method and apparatus for frame-based loudspeaker equalization 失效
    用于基于帧的扬声器均衡的方法和装置

    公开(公告)号:US20060133620A1

    公开(公告)日:2006-06-22

    申请号:US11312009

    申请日:2005-12-19

    申请人: Khosrow Lashkari

    发明人: Khosrow Lashkari

    IPC分类号: H04R29/00 H04R3/00

    CPC分类号: H04R3/04

    摘要: A method and apparatus for loudpeaker equalization are described. In one embodiment, the method comprising generating a set of parameters using an invertible, non-linear system based on input audio data and output data corresponding to a prediction of an output of a loudspeaker in response to the input data, and controlling an exact non-linear inverse of the non-linear system using the set of parameters to output a predistorted version of the input data.

    摘要翻译: 描述了用于扬声器均衡的方法和装置。 在一个实施例中,该方法包括使用基于输入音频数据的可逆非线性系统和对应于响应于输入数据的扬声器的输出的预测的输出数据来生成一组参数,并且控制精确的非线性 使用该参数集的非线性系统的线性反演来输出输入数据的预失真版本。

    Method and apparatus for frame-based loudspeaker equalization
    4.
    发明授权
    Method and apparatus for frame-based loudspeaker equalization 失效
    用于基于帧的扬声器均衡的方法和装置

    公开(公告)号:US07826625B2

    公开(公告)日:2010-11-02

    申请号:US11312009

    申请日:2005-12-19

    申请人: Khosrow Lashkari

    发明人: Khosrow Lashkari

    IPC分类号: H04R3/00

    CPC分类号: H04R3/04

    摘要: A method and apparatus for loudspeaker equalization are described. In one embodiment, the method comprising generating a set of parameters using an invertible, non-linear system based on input audio data and output data corresponding to a prediction of an output of a loudspeaker in response to the input data, and controlling an exact non-linear inverse of the non-linear system using the set of parameters to output a predistorted version of the input data.

    摘要翻译: 描述用于扬声器均衡的方法和装置。 在一个实施例中,该方法包括使用基于输入音频数据的可逆非线性系统和对应于响应于输入数据的扬声器的输出的预测的输出数据来生成一组参数,并且控制精确的非线性 使用该参数集的非线性系统的线性反演来输出输入数据的预失真版本。

    Energy-based nonuniform time-scale modification of audio signals
    5.
    发明授权
    Energy-based nonuniform time-scale modification of audio signals 失效
    基于能量的不均匀时间尺度修改音频信号

    公开(公告)号:US07426470B2

    公开(公告)日:2008-09-16

    申请号:US10264042

    申请日:2002-10-03

    IPC分类号: G10L21/04

    CPC分类号: G10L21/04

    摘要: A method for energy based, non-uniform time-scale compression of audio signals includes receiving a frame of data corresponding to an input audio signal and segmenting the data into a plurality of segments. The method further includes estimating a value related to energy of the frame of data, determining a peak energy estimate for the frame, determining an energy threshold based on the peak energy estimate of the frame and comparing the value related to energy of the frame of the data with the energy threshold to control time-scale compression of the audio data.

    摘要翻译: 用于音频信号的基于能量的,不均匀的时间尺度压缩的方法包括接收对应于输入音频信号的数据帧并将数据分割成多个段。 该方法还包括估计与数据帧的能量有关的值,确定帧的峰值能量估计,基于帧的峰值能量估计确定能量阈值,并将与帧的能量相关的值进行比较 具有能量阈值的数据以控制音频数据的时间尺度压缩。

    Gradient descent optimization of linear prediction coefficients for speech coders
    6.
    发明授权
    Gradient descent optimization of linear prediction coefficients for speech coders 有权
    语音编码器线性预测系数的梯度下降优化

    公开(公告)号:US07200552B2

    公开(公告)日:2007-04-03

    申请号:US10134281

    申请日:2002-04-29

    IPC分类号: G10L19/12

    摘要: An optimization algorithm is provided for linear prediction based speech coding systems. The optimization algorithm minimizes the error between original speech samples and synthesized speech samples. Optimized linear prediction coefficients are computed directly from a system difference equation without converting the coefficients into the root-domain.

    摘要翻译: 为基于线性预测的语音编码系统提供了优化算法。 优化算法使原始语音样本和合成语音样本之间的误差最小化。 优化的线性预测系数直接从系统差分方程计算,而不将系数转换成根域。

    Joint optimization of excitation and model parameters in parametric speech coders
    7.
    发明授权
    Joint optimization of excitation and model parameters in parametric speech coders 有权
    参数语音编码器中激励和模型参数的联合优化

    公开(公告)号:US06859775B2

    公开(公告)日:2005-02-22

    申请号:US09800071

    申请日:2001-03-06

    IPC分类号: G10L19/00 G10L19/12 G01L19/04

    CPC分类号: G10L19/12 G10L2019/0013

    摘要: A speech synthesis system is provided that optimizes a synthesis filter. Optimization is achieved by minimizing a synthesis error between the original speech sample and a synthesized speech sample. A gradient search algorithm in the root domain is also provided to aid minimization of the synthesis error.

    摘要翻译: 提供了优化合成滤波器的语音合成系统。 通过最小化原始语音样本和合成语音样本之间的合成误差来实现优化。 还提供了根域中的梯度搜索算法以帮助最小化合成误差。

    Inductive energization system and method for vehicles
    8.
    发明授权
    Inductive energization system and method for vehicles 失效
    车辆感应通电系统及方法

    公开(公告)号:US5207304A

    公开(公告)日:1993-05-04

    申请号:US801743

    申请日:1991-12-03

    IPC分类号: B60L9/00

    摘要: An inductive energization system for moving vehicles includes wayside inductors under the roadway and pickup inductor circuits in electrically powered vehicles. A pickup power controller has a switching circuit, including a zero-crossing trigger circuit, a current limiting inductor, and a bleed resistor. The controller provides for fast switching, desirable for closed loop control of the inductive energy transfer system, as well as low harmonic distortion of waveforms, low acoustic noise, and low maintenance requirements. The pickup inductor of the preferred embodiment has rigid metal conductors bonded together into a single member, allowing this element to serve as both a current carrying element as well as a primary structural member of the pickup inductor. The roadway inductor is split into many segments. Sensors in the roadway detect when vehicles requiring power are present, and a wayside inductor segment controller responds to the sensory signals by energizing only those wayside inductor segments needed to transfer power to such vehicles. This control methodology improves the energy efficiency of the system. In addition, the roadway sensors can be designed to detect identification signals broadcast by vehicle identification transmitters, thereby enabling the system to charge for energy usage by each vehicle.

    摘要翻译: 用于移动车辆的感应通电系统包括道路下的路旁电感和电动车辆中的拾音器电感电路。 拾取功率控制器具有开关电路,包括过零触发电路,限流电感器和放电电阻器。 控制器提供快速切换,对于感应能量传输系统的闭环控制以及波形的低谐波失真,低噪声和低维护要求是理想的。 优选实施例的拾取电感器具有将金属导体接合在一起的单个构件,允许该元件既作为载流元件又作为拾取电感器的主要构件。 道路电感分为许多段。 巷道中的传感器检测到需要电力的车辆是否存在,并且旁路电感器段控制器仅通过向仅向这些车辆传递动力所需的那些路旁的电感器段通电来响应感觉信号。 该控制方法提高了系统的能源效率。 此外,道路传感器可以被设计为检测由车辆识别发射机广播的识别信号,从而使得系统能够对每个车辆的能量使用进行收费。

    Modified volterra-wiener-hammerstein (MVWH) method for loudspeaker modeling and equalization
    9.
    发明授权
    Modified volterra-wiener-hammerstein (MVWH) method for loudspeaker modeling and equalization 失效
    用于扬声器建模和均衡的改进的volterra-wiener-hammerstein(MVWH)方法

    公开(公告)号:US07873172B2

    公开(公告)日:2011-01-18

    申请号:US11446085

    申请日:2006-05-31

    申请人: Khosrow Lashkari

    发明人: Khosrow Lashkari

    IPC分类号: H04R29/00

    摘要: A method and apparatus for adaptive precompensation is disclosed. In one embodiment, the method comprises modifying operation of a predistortion filter in response to previous predistorted values and an original input signal, determining a precompensation error between the original input samples and the predicted loudspeaker output, and substantially reducing the precompensation error by using the exact inverse of a loudspeaker model that is a cascaded arrangement of at least one linear system with a non-linear system.

    摘要翻译: 公开了一种用于自适应预补偿的方法和装置。 在一个实施例中,该方法包括响应于先前的预失真值和原始输入信号修改预失真滤波器的操作,确定原始输入样本与预测的扬声器输出之间的预补偿误差,并且通过使用确切的 作为具有非线性系统的至少一个线性系统的级联布置的扬声器模型的倒数。

    ENERGY-BASED NONUNIFORM TIME-SCALE MODIFICATION OF AUDIO SIGNALS

    公开(公告)号:US20080133252A1

    公开(公告)日:2008-06-05

    申请号:US11971625

    申请日:2008-01-09

    IPC分类号: G10L21/04

    CPC分类号: G10L21/04

    摘要: A method for energy based, non-uniform time-scale compression of audio signals includes receiving a frame of data corresponding to an input audio signal and segmenting the data into a plurality of segments. The method further includes estimating a value related to energy of the frame of data, determining a peak energy estimate for the frame, determining an energy threshold based on the peak energy estimate of the frame and comparing the value related to energy of the frame of the data with the energy threshold to control time-scale compression of the audio data.