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公开(公告)号:US20070038444A1
公开(公告)日:2007-02-15
申请号:US11362286
申请日:2006-02-23
申请人: Markus Buck , Gerhard Schmidt , Tim Haulick
发明人: Markus Buck , Gerhard Schmidt , Tim Haulick
CPC分类号: B60R16/0373 , G10L2021/02165
摘要: A system for automatic adjustment of at least one element provided with at least one actuator and provided inside or outside a vehicular cabin is provided. The system includes at least one microphone array for detecting utterances of at least one speaker and for obtaining microphone signals corresponding to the utterances. A localization detector is configured to determine the position of the speaker on the basis of the microphone signals. The localization detector outputs a localization signal to a controller. The controller is configured to receive the localization signal and automatically controls the actuator of the element on the basis of the localization signal. A method for automatically adjusting at least one element provided with at least one actuator and provided inside or outside a vehicular cabin is also provided. The method includes detecting utterances of a speaker by a microphone array and obtaining microphone signals corresponding to the utterances. The method further includes determining the position of the speaker on the basis of the microphone signals and automatically controlling the actuator of the element on the basis of the determined position of the speaker to adjust the element.
摘要翻译: 提供了一种用于自动调节至少一个设置有至少一个致动器并且设置在车厢内部或外部的元件的系统。 该系统包括至少一个麦克风阵列,用于检测至少一个扬声器的话语,并且用于获得对应于话语的麦克风信号。 定位检测器被配置为基于麦克风信号确定扬声器的位置。 定位检测器向控制器输出定位信号。 控制器被配置为接收定位信号并且基于定位信号自动地控制元件的致动器。 还提供一种用于自动调节至少一个设置有至少一个致动器并且设置在车厢内部或外部的元件的方法。 该方法包括通过麦克风阵列检测扬声器的话语并获得对应于话语的麦克风信号。 该方法还包括基于麦克风信号来确定扬声器的位置,并且基于所确定的扬声器的位置来自动控制元件的致动器以调整元件。
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公开(公告)号:US20080107280A1
公开(公告)日:2008-05-08
申请号:US11924987
申请日:2007-10-26
申请人: Tim Haulick , Gerhard Schmidt
发明人: Tim Haulick , Gerhard Schmidt
IPC分类号: H04B3/20
CPC分类号: H04R3/005 , G10L2021/02082 , G10L2021/02165 , G10L2021/02166 , H04M9/082 , H04R2410/05 , H04R2430/23 , H04R2430/25 , H04R2499/13
摘要: A communication system enhances communications in a noisy environment. Multiple input arrays convert a voiced or unvoiced signal into an analog signal. A converter receives the analog signal and generates digital signals. A digital signal processor determines temporal and spatial information from the digital signals. The processed signals are then converted to audible sound.
摘要翻译: 通信系统在嘈杂的环境中增强通信。 多输入阵列将有声或无声信号转换为模拟信号。 A转换器接收模拟信号并产生数字信号。 数字信号处理器根据数字信号确定时间和空间信息。 经处理的信号然后被转换为可听见的声音。
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公开(公告)号:US20080031467A1
公开(公告)日:2008-02-07
申请号:US11788069
申请日:2007-04-18
申请人: Tim Haulick , Gerhard Schmidt , Martin Roessler
发明人: Tim Haulick , Gerhard Schmidt , Martin Roessler
IPC分类号: H04B3/20
CPC分类号: H04M9/082
摘要: The present invention relates to a method for reducing an echo in a microphone signal generated by a microphone, comprising echo compensating the microphone signal by subtracting an estimated echo signal from the microphone signal to generate an echo compensated signal, detecting a speech activity of a local speaker on the basis of the microphone signal and the estimated echo signal and suppressing a residual echo in the echo compensated signal on the basis of the detected speech activity to obtain an output signal. The invention further relates to a system for processing a microphone signal generated by a microphone, comprising echo compensation filtering means configured to receive and echo compensate the microphone signal to output an echo compensated signal based on the received microphone signal, a speech activity detection means configured to detect speech activity of a local speaker by receiving and analyzing the echo compensated signal and to output a detection signal and a residual echo suppressing means configured to receive the detection signal and to receive and filter the echo compensated signal on the basis of the detection signal to output an output signal.
摘要翻译: 本发明涉及一种用于减少由麦克风产生的麦克风信号中的回波的方法,包括通过从麦克风信号中减去估计的回波信号来补偿麦克风信号的回波,以产生回波补偿信号,检测本地的话音活动 基于所述麦克风信号和所述估计回波信号,基于检测到的语音活动抑制所述回波补偿信号中的残留回波,以获得输出信号。 本发明还涉及一种用于处理由麦克风产生的麦克风信号的系统,包括:回波补偿滤波装置,被配置为接收和回波补偿麦克风信号,以基于接收到的麦克风信号输出回波补偿信号;配置的语音活动检测装置 通过接收和分析回波补偿信号来检测本地扬声器的语音活动,并输出检测信号和残余回声抑制装置,被配置为接收检测信号,并且基于检测信号接收和滤波回波补偿信号 输出输出信号。
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公开(公告)号:US20080031466A1
公开(公告)日:2008-02-07
申请号:US11787348
申请日:2007-04-16
申请人: Markus Buck , Tim Haulick , Martin Roessler , Gerhard Schmidt
发明人: Markus Buck , Tim Haulick , Martin Roessler , Gerhard Schmidt
IPC分类号: H04B3/20
CPC分类号: H04M9/082 , H04R3/02 , H04R2499/13
摘要: The invention is directed to a multi-channel echo compensation system, comprising two loudspeaker input channels, each loudspeaker input channel being connected to a loud-speaker for providing a loudspeaker input signal to be emanated by the loudspeaker, a microphone output channel being connected to at least one microphone for receiving a microphone output signal from the at least one microphone, wherein each microphone is configured to acquire a signal emanating from the loudspeakers, a compensation channel for each loudspeaker input channel, each compensation channel connecting a respective loudspeaker input channel and the microphone output channel, an adaptive compensation filter for each compensation channel, wherein each adaptive compensation filter is configured to filter a signal on the respective compensation channel such that a compensation output signal is provided to compensate a microphone output signal for a signal emanating from the loudspeakers, a pre-processing means for pre-processing loudspeaker input signals on the compensation channels, the pre-processing means being configured to determine a correlation value of the loudspeaker input signals for the two loudspeakers according to a pre-determined criterion and to de-activate one of the adaptive compensation filters if the determined correlation value passes a pre-determined threshold.
摘要翻译: 本发明涉及一种多通道回波补偿系统,包括两个扬声器输入通道,每个扬声器输入通道连接到扬声器,用于提供由扬声器发出的扬声器输入信号;麦克风输出通道连接到 至少一个麦克风,用于从所述至少一个麦克风接收麦克风输出信号,其中每个麦克风被配置为获取从扬声器发出的信号,每个扬声器输入通道的补偿通道,每个补偿通道连接相应的扬声器输入通道和 麦克风输出通道,用于每个补偿通道的自适应补偿滤波器,其中每个自适应补偿滤波器被配置为对相应补偿通道上的信号进行滤波,使得提供补偿输出信号以补偿麦克风输出信号,以产生从 扬声器,预处理手段 用于在补偿通道上预处理扬声器输入信号,预处理装置被配置为根据预定准则确定两个扬声器的扬声器输入信号的相关值,并且去激活自适应补偿中的一个 如果确定的相关值通过预定阈值,则滤波器。
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公开(公告)号:US20050265560A1
公开(公告)日:2005-12-01
申请号:US11118092
申请日:2005-04-29
申请人: Tim Haulick , Gerhard Schmidt
发明人: Tim Haulick , Gerhard Schmidt
IPC分类号: H04R1/40 , G10K11/178 , G10K15/00 , H03G5/00 , H03G5/16 , H03H21/00 , H04R3/00 , H04R3/04 , H04R5/02 , H04R27/00 , H04R29/00
CPC分类号: H03G5/165 , H04R3/02 , H04R5/02 , H04R29/001 , H04R2499/13
摘要: A system automatically determines an equalizing filter characteristic for a communication system within a vehicle. The communication system includes a loudspeaker and a microphone or microphone array. The system transmits a predetermined test signal through the loudspeaker and receive the test signal through the microphone or microphone array. Based on the predetermined test signal and the received test signal, a transfer function is developed. The equalizing filter characteristic is then developed from the transfer function.
摘要翻译: 系统自动确定车辆内的通信系统的均衡滤波器特性。 通信系统包括扬声器和麦克风或麦克风阵列。 系统通过扬声器发送预定的测试信号,并通过麦克风或麦克风阵列接收测试信号。 基于预定的测试信号和接收到的测试信号,开发传递函数。 然后从传递函数开发均衡滤波器特性。
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16.
公开(公告)号:US08705753B2
公开(公告)日:2014-04-22
申请号:US12174595
申请日:2008-07-16
申请人: Tim Haulick , Viola Hoff , Gerhard Schmidt
发明人: Tim Haulick , Viola Hoff , Gerhard Schmidt
IPC分类号: H03G3/24
CPC分类号: H04M9/082 , G10L2021/02082
摘要: The present invention relates to a method for processing sound signals in a vehicle multimedia system. A sound signal is recorded by a microphone. A voice signal component resulting from reproducing a voice signal and an audio signal component resulting from reproducing an audio signal are compensated in the recorded sound signal. The invention further relates to a system for processing sound signals and a vehicle multimedia system.
摘要翻译: 本发明涉及一种用于处理车载多媒体系统中的声音信号的方法。 麦克风记录声音信号。 在记录的声音信号中补偿由再现音频信号而产生的语音信号和音频信号分量产生的语音信号分量。 本发明还涉及一种用于处理声音信号的系统和车载多媒体系统。
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公开(公告)号:US08306234B2
公开(公告)日:2012-11-06
申请号:US11753255
申请日:2007-05-24
申请人: Markus Christoph , Tim Haulick , Gerhard Schmidt
发明人: Markus Christoph , Tim Haulick , Gerhard Schmidt
IPC分类号: H04R5/00
CPC分类号: H04R3/02 , G10K11/178 , G10K2210/503 , G10K2210/506 , H04R2499/13
摘要: Improving the acoustical communication between interlocutors in at least two positions in a room includes generating electrical signals representative of acoustical signals present at the respective interlocutor positions, amplifying each of the electrical signals and converting the amplified electrical signals into acoustical signals. A time delay is applied to the electrical signals such that the acoustical signal arriving first at one of the interlocutor positions originates from the direction of the other interlocutor position.
摘要翻译: 改善房间中至少两个位置的对话者之间的声学通信包括产生表示存在于相应对话者位置的声信号的电信号,放大每个电信号并将放大的电信号转换成声信号。 对电信号施加时间延迟,使得首先到达对讲机位置之一的声信号来自另一个对讲机位置的方向。
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18.
公开(公告)号:US20100150364A1
公开(公告)日:2010-06-17
申请号:US12636160
申请日:2009-12-11
申请人: Markus Buck , Tobias Wolff , Gerhard Schmidt , Tim Haulick
发明人: Markus Buck , Tobias Wolff , Gerhard Schmidt , Tim Haulick
CPC分类号: G01S3/807 , G10L21/02 , H04M9/082 , H04R1/406 , H04R3/005 , H04R3/02 , H04R2410/01 , H04R2430/23 , H04R2430/25 , H04R2499/13 , H04S1/00 , H04S3/00
摘要: The invention provides a computer-implemented method for determining a time delay for time delay compensation of a microphone signal from a microphone array in a beamformer arrangement. For a given time, an instantaneous estimate of a position of a wanted sound source and/or of a direction of arrival of a signal originating from the wanted sound source is determined. The computer system then determines whether the instantaneous estimate deviates from a preset estimate of a position of the wanted sound source and/or of a direction of arrival of a signal originating from the wanted sound source according to a predetermined criterion. The predetermined criterion comprises a check whether the instantaneous estimate deviates from the preset estimate by at least a predetermined deviation threshold. If the predetermined criterion is fulfilled, the instantaneous estimate for the given time is set by the computer system as the preset estimate, and the computer system determines the time delay for time delay compensation of the microphone signal based on the instantaneous estimate.
摘要翻译: 本发明提供了一种计算机实现的方法,用于确定来自波束形成器布置中的麦克风阵列的麦克风信号的时间延迟的时间延迟。 对于给定时间,确定所需声源的位置和/或源自所需声源的信号的到达方向的瞬时估计。 计算机系统然后根据预定标准确定瞬时估计是否偏离预期的所需声源的位置的预设估计和/或源自所需声源的信号的到达方向。 预定标准包括检查瞬时估计是否偏离预设估计至少预定的偏差阈值。 如果满足预定标准,则由计算机系统将给定时间的瞬时估计值设置为预设估计,并且计算机系统基于瞬时估计确定麦克风信号的时间延迟补偿的时间延迟。
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公开(公告)号:US20100036659A1
公开(公告)日:2010-02-11
申请号:US12537749
申请日:2009-08-07
申请人: Tim Haulick , Mohamed Krini , Shreyas Paranjpe , Gerhard Schmidt
发明人: Tim Haulick , Mohamed Krini , Shreyas Paranjpe , Gerhard Schmidt
CPC分类号: G10L19/012 , G10L21/0208
摘要: The present invention relates to a method for signal processing comprising the steps of providing a set of prototype spectral envelopes, providing a set of reference noise prototypes, wherein the reference noise prototypes are obtained from at least a sub-set of the provided set of prototype spectral envelopes, detecting a verbal utterance by at least one microphone to obtain a microphone signal, processing the microphone signal for noise reduction based on the provided reference noise prototypes to obtain an enhanced signal and encoding the enhanced signal based on the provided prototype spectral envelopes to obtain an encoded enhanced signal.
摘要翻译: 本发明涉及一种用于信号处理的方法,包括以下步骤:提供一组原型频谱包络,提供一组参考噪声原型,其中参考噪声原型从所提供的一组原型的至少一个子集获得 通过至少一个麦克风检测语音话语以获得麦克风信号,基于所提供的参考噪声原型处理麦克风信号以进行噪声降低,以获得增强信号,并基于所提供的原型频谱包络对增强信号进行编码, 获得编码的增强信号。
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公开(公告)号:US20070110254A1
公开(公告)日:2007-05-17
申请号:US11411004
申请日:2006-04-25
申请人: Markus Christoph , Tim Haulick , Gerhard Schmidt
发明人: Markus Christoph , Tim Haulick , Gerhard Schmidt
IPC分类号: H04B3/20
CPC分类号: H04M3/56 , G10L2021/02082 , H04M9/082 , H04R3/005 , H04R3/02 , H04R3/12 , H04R2499/13
摘要: A dereverberation and feedback compensation system reduces the echo received by a first audio device while reducing the speech feedback received from a second audio device. A decorrelation logic decorrelates audio signals from the first audio device. A first processor generates a noise compensation signal based on the decorrelated audio signals and system determined filter coefficients. The second processor generates an enhanced noise correlation signal based on speech signals of a second audio device and the filter coefficients used by the first processor.
摘要翻译: 去混响和反馈补偿系统减少由第一音频设备接收的回声,同时减少从第二音频设备接收的语音反馈。 解相关逻辑去除了来自第一音频设备的音频信号。 第一处理器基于去相关音频信号和系统确定的滤波器系数产生噪声补偿信号。 第二处理器基于第二音频设备的语音信号和由第一处理器使用的滤波器系数产生增强的噪声相关信号。
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