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1.发明授权公开(公告)号：US07912228B2
公开(公告)日：2011-03-22
申请号：US10623286
申请日：2003-07-18
CPC分类号： H04R3/02 , H04R2499/13
摘要： In a method and equipment for operating a voice-supported system, such as a communications and/or intercom/two-way intercom device in a motor vehicle, using at least one microphone and at least one loudspeaker to reproduce a signal generated by the microphone, as well as a bandpass filter configured between the microphone and the loudspeaker, a power of the signal as a function of a frequency is determined, and the bandpass filter is adjusted as a function of at least one local maximum of the power of the signal as a function of the frequency.
摘要翻译： 在用于操作诸如机动车辆中的通信和/或对讲/双向对讲设备的语音支持系统的方法和设备中，使用至少一个麦克风和至少一个扬声器来再现由麦克风产生的信号 以及配置在麦克风和扬声器之间的带通滤波器，确定作为频率的函数的信号的功率，并且根据信号的功率的至少一个局部最大值来调整带通滤波器 作为频率的函数。
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公开(公告)号：US07467084B2
公开(公告)日：2008-12-16
申请号：US10360889
申请日：2003-02-07
IPC分类号： G10L21/00
CPC分类号： G10L21/0208 , H04R3/005 , H04R3/02 , H04R2499/13
摘要： A method and a device are for operating a voice-enhancement system, such as a communication and/or intercom/two-way intercom or duplex telephony device in a motor vehicle. The device includes at least one microphone and at least one loudspeaker for reproducing a signal generated by the microphone, as well as a bandpass filter configured between the microphone and the loudspeaker. The bandpass filter is adjusted as a function of a comparison between the power of the signal generated by the microphone at a test frequency, and the power of the signal generated by the microphone at an at least substantially integral multiple of the test frequency, or as a function of a comparison between the power of the signal generated by the microphone at a test frequency, and the power of the signal generated by the microphone at the test frequency at at least an earlier point in time.
摘要翻译： 一种方法和设备用于操作诸如机动车辆中的通信和/或对讲/双向对讲或双工电话设备的语音增强系统。 该装置包括至少一个麦克风和用于再现由麦克风产生的信号的至少一个扬声器，以及配置在麦克风和扬声器之间的带通滤波器。 根据在测试频率下由麦克风产生的信号的功率与麦克风在测试频率的至少基本上整数倍产生的信号的功率之间的比较，或者如 在测试频率下由麦克风产生的信号的功率与麦克风在测试频率至少较早的时间点产生的信号的功率之间的比较的功能。
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公开(公告)号：US5710822A
公开(公告)日：1998-01-20
申请号：US553186
申请日：1995-11-07
IPC分类号： G10K11/178 , G10K11/16
CPC分类号： G10K11/1784 , G10K2210/117 , G10K2210/3028 , G10K2210/3039 , G10K2210/3049
摘要： An active adaptive control system and method has frequency dependent filtering with a transfer characteristic which is a function of a frequency dependent shaped power limitation characteristic maximizing usage of available output transducer authority. Band separation is provided for different tones. Power limit partitioning is provided for effectively distributing power between correction tones to maximize model performance.
摘要翻译： 主动自适应控制系统和方法具有频率相关滤波，其具有传递特性，其是频率依赖的形状功率限制特性的函数，其最大化可用输出传感器权威的使用。 为不同的音调提供频带分离。 提供功率限制分区，用于在校正音之间有效分配功率，以最大限度地提高模型性能。
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公开(公告)号：US5278913A
公开(公告)日：1994-01-11
申请号：US920774
申请日：1992-07-28
IPC分类号： G10K11/178 , G10K11/16
CPC分类号： G10K11/1784 , G10K2210/3017 , G10K2210/3039 , G10K2210/3045 , G10K2210/3049 , G10K2210/3222 , G10K2210/3228
摘要： An active acoustic attenuation system prevents overdriving of the canceling output transducer or speaker (14) by shunting at least part of the correction signal (46) to a parallel shunt path (306) and away from the output transducer (14) Variable gains (308 and 310) are provided in the shunt path (306) and the input to the output transducer (14) for varying the ratio between the part of the correction signal (46) supplied to the output transducer (14) and the part of the correction signal (46) shunted to the shunt path (306). A first adaptive filter model (40) has an error input (202) from the error signal and outputs the correction signal (46). A second adaptive filter model (142) models the output transducer (14) and the error path (56) between the output transducer (14) and the error transducer (16). A copy (312) of the second model (142) has an input from the output (46) of the first model (40), and the output of the copy (312) is summed with the error signal (44) and the resultant sum is supplied to the error input (202) of the first model (40), such that the shunt path (306) is provided through the model copy (312).
摘要翻译： 主动声衰减系统通过将至少部分校正信号（46）分流到并联分流路径（306）并远离输出换能器（14）来防止消除输出换能器或扬声器（14）的过驱动。可变增益（308） 和310）设置在分流路径（306）中，并且输入到输出变换器（14），用于改变提供给输出变换器（14）的校正信号（46）的部分与校正信号的部分之间的比率 信号（46）分流到分流路径（306）。 第一自适应滤波器模型（40）具有来自误差信号的误差输入（202）并输出校正信号（46）。 第二自适应滤波器模型（142）对输出变换器（14）和输出变换器（14）和误差变换器（16）之间的误差路径（56）进行建模。 第二模型（142）的副本（312）具有来自第一模型（40）的输出（46）的输入，并且复制（312）的输出与误差信号（44）相加，并且所得到的 总和被提供给第一模型（40）的误差输入（202），使得分流路径（306）通过模型拷贝（312）提供。
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公开(公告)号：US07434205B1
公开(公告)日：2008-10-07
申请号：US11061113
申请日：2005-02-18
IPC分类号： G06F9/44
CPC分类号： G06F8/20 , G06F9/451 , G06F11/3664
摘要： Interpretation of virtual types in a software development, debugging, or monitoring environment. Display and modification of variables having virtual types. Detection of virtual types.
摘要翻译： 在软件开发，调试或监控环境中解释虚拟类型。 显示和修改具有虚拟类型的变量。 检测虚拟类型。
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公开(公告)号：US06665411B2
公开(公告)日：2003-12-16
申请号：US09790410
申请日：2001-02-21
申请人： Shawn K. Steenhagen
发明人： Shawn K. Steenhagen
IPC分类号： H04R2700
摘要： A digital voice enhancement, DVE, communication system includes an instability detector detecting an unstable acoustic feedback condition from a loudspeaker to a microphone by sensing a condition of the electrical signal transmitted from the microphone to the loudspeaker, and a corrective processor responsive to the instability detector to modify the electrical signal to reduce unstable acoustic feedback. The sensed condition may be magnitude, power, or, preferably, the sinusoidal characteristic of the electrical signal, namely the electrical signal becoming sinusoidal in nature.
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公开(公告)号：US06549629B2
公开(公告)日：2003-04-15
申请号：US09790408
申请日：2001-02-21
申请人： Brian M. Finn , Shawn K. Steenhagen
发明人： Brian M. Finn , Shawn K. Steenhagen
IPC分类号： H04R3100
CPC分类号： H04R3/005
摘要： In a DVE, digital voice enhancement, communication system, the selection decision for choosing which microphone to be active is based on a given function of the speech of a respective talker relative to his/her acoustic environment at the respective microphone. The selection decision is based on a selection technique normalizing at least one of a) different microphone sensitivities and b) different background noise levels at the respective microphones, preferably based on the ratio of how much louder a talker speaks over the background noise at his/her respective microphone.
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