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

IPC分类号： H04B3/20 ,  H04R3/00

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。

公开(公告)号：US20100014690A1

公开(公告)日：2010-01-21

申请号：US12504333

申请日：2009-07-16

申请人： Tobias Wolff ,  Markus Buck ,  Gerhard Schmidt

发明人： Tobias Wolff ,  Markus Buck ,  Gerhard Schmidt

IPC分类号： H04R3/00

CPC分类号： H04R3/005 ,  H04R2430/20 ,  H04R2430/23

摘要： Embodiments of the present invention relate to methods, systems, and computer program products for signal processing. A first plurality of microphone signals is obtained by a first microphone array. A second plurality of microphone signals is obtained by a second microphone array different from the first microphone array. The first plurality of microphone signals is beamformed by a first beamformer comprising beamforming weights to obtain a first beamformed signal. The second plurality of microphone signals is beamformed by a second beamformer comprising the same beamforming weights as the first beamformer to obtain a second beamformed signal. The beamforming weights are adjusted such that the power density of echo components and/or noise components present in the first and second plurality of microphone signals is substantially reduced.

摘要翻译： 本发明的实施例涉及用于信号处理的方法，系统和计算机程序产品。 第一麦克风信号由第一麦克风阵列获得。 通过与第一麦克风阵列不同的第二麦克风阵列获得第二多个麦克风信号。 第一组多个麦克风信号由包括波束成形权重的第一波束形成器波束形成，以获得第一波束形成信号。 第二组麦克风信号由包括与第一波束形成器相同的波束形成权重的第二波束形成器波束形成，以获得第二波束形成信号。 调整波束成形权重使得第一和第二多个麦克风信号中存在的回波分量和/或噪声分量的功率密度显着降低。

公开(公告)号：US20060253282A1

公开(公告)日：2006-11-09

申请号：US11376001

申请日：2006-03-14

申请人： Gerhard Schmidt ,  Markus Buck ,  Tim Haulick

发明人： Gerhard Schmidt ,  Markus Buck ,  Tim Haulick

IPC分类号： G10L15/20

CPC分类号： G07C5/0808

摘要： A system automatically recognizes a vehicle operating condition through a microphone positioned within the vehicle. The microphone detects acoustic signals. A database stores speech templates and operating noise templates. A feature extracting module receives microphone signals and extracts a set of operating noise feature parameters or speech feature parameters from the microphone signals. A speech and noise recognition module may determine an operating noise template that best matches a set of extracted operating noise feature parameters and/or a speech template. The speech template best matches the set of extracted speech feature parameters.

摘要翻译： 系统通过位于车辆内的麦克风自动识别车辆操作状态。 麦克风检测声信号。 数据库存储语音模板和操作噪声模板。 特征提取模块从麦克风信号接收麦克风信号并提取一组操作噪声特征参数或语音特征参数。 语音和噪声识别模块可以确定与提取的操作噪声特征参数和/或语音模板的一组最佳匹配的操作噪声模板。 语音模板最适合提取的语音特征参数集合。

公开(公告)号：US20080031471A1

公开(公告)日：2008-02-07

申请号：US11749678

申请日：2007-05-16

申请人： Tim Haulick ,  Gerhard Schmidt ,  Markus Buck

发明人： Tim Haulick ,  Gerhard Schmidt ,  Markus Buck

IPC分类号： H04B1/00

CPC分类号： H03G3/32 ,  H03G5/16 ,  H04R3/04 ,  H04R2499/13

摘要： An equalization system enhances the quality of communications between a remote party and a local party. The equalization system includes an equalization filter that equalizes an acoustic signal received from the remote party. The equalized acoustic signal is transmitted to a speaker based on the equalized acoustic signal. A device converts sound into electrical signals. The electrical signals are transmitted to an echo compensation filter that compensates for reflected sound. Filter characteristics of the equalization filter are based on filter characteristics of the echo compensation filter.

摘要翻译： 均衡系统提高了远程方与本地方之间的通信质量。 均衡系统包括均衡滤波器，其均衡从远程方接收的声信号。 均衡的声信号基于均衡的声信号被发送到扬声器。 设备将声音转换为电信号。 电信号被传送到补偿反射声音的回波补偿滤波器。 均衡滤波器的滤波特性基于回波补偿滤波器的滤波特性。

公开(公告)号：US20080031469A1

公开(公告)日：2008-02-07

申请号：US11789798

申请日：2007-04-25

申请人： Tim Haulick ,  Gerhard Schmidt ,  Markus Buck ,  Martin Roessler ,  Hans-Joerg Koepf

发明人： Tim Haulick ,  Gerhard Schmidt ,  Markus Buck ,  Martin Roessler ,  Hans-Joerg Koepf

IPC分类号： H04R3/02

CPC分类号： H04M9/082

摘要： A system reduces echoes in an audio system by de-correlating audio signals. Adaptive filters provide compensation based on the de-correlated signals. A controller controls the de-correlation based upon the adaptation state of the adaptive filters.

摘要翻译： 系统通过使音频信号去相关来减少音频系统中的回波。 自适应滤波器基于去相关信号提供补偿。 控制器基于自适应滤波器的适应状态来控制去相关。

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

IPC分类号： H04R3/00 ,  G10L15/26

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。

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。

公开(公告)号：US20070280486A1

公开(公告)日：2007-12-06

申请号：US11740164

申请日：2007-04-25

申请人： Markus Buck ,  Tim Haulick ,  Gerhard Schmidt

发明人： Markus Buck ,  Tim Haulick ,  Gerhard Schmidt

IPC分类号： H04R3/00

CPC分类号： H04R27/00 ,  H04R2499/13 ,  H04S7/301 ,  H04S7/303

摘要： The present invention relates to a vehicle communication system comprising a plurality of microphones adapted to detect speech signals of different vehicle passengers, a mixer combining the audio signal components of the different microphones to a resulting speech output signal, a weighting unit determining the weighting of the audio signal components for the resulting speech output signal, where the weighting unit determines the weighting of the signal components based upon non-acoustical information about the presence of a vehicle passenger.

摘要翻译： 车载通信系统技术领域本发明涉及一种车辆通信系统，其特征在于，包括：适合于检测不同车辆乘客的语音信号的多个麦克风，混合不同麦克风的音频信号成分到合成语音输出信号的混合器;加权单元， 用于所得语音输出信号的音频信号分量，其中加权单元基于关于车辆乘客的存在的非声学信息来确定信号分量的加权。

公开(公告)号：US20070135061A1

公开(公告)日：2007-06-14

申请号：US11492675

申请日：2006-07-25

申请人： Markus Buck ,  Tim Haulick ,  Gerhard Schmidt ,  Michael Tropp

发明人： Markus Buck ,  Tim Haulick ,  Gerhard Schmidt ,  Michael Tropp

IPC分类号： H04B1/38 ,  H04B1/034

CPC分类号： H04M1/6083 ,  H04M9/001 ,  H04M2250/02 ,  H04R2420/07 ,  H04R2499/13 ,  H04W4/04 ,  H04W88/02

摘要： A vehicle communication system detects the presence of a passenger wearable communication device. The system receives audio signals from multiple sources inside or outside of a vehicle. The system processes the signals before routing the signals to multiple destinations. The destinations may include wearable personal communication devices, front and/or rear speakers, and/or a remote mobile device.

摘要翻译： 车辆通信系统检测乘客可穿戴通信设备的存在。 该系统从车辆内部或外部的多个源接收音频信号。 在将信号路由到多个目的地之前，系统处理信号。 目的地可以包括穿戴式个人通信设备，前置和/或后置扬声器和/或远程移动设备。

公开(公告)号：US09076455B2

公开(公告)日：2015-07-07

申请号：US13591667

申请日：2012-08-22

申请人： Mohamed Krini ,  Gerhard Schmidt ,  Bernd Iser ,  Arthur Wolf

发明人： Mohamed Krini ,  Gerhard Schmidt ,  Bernd Iser ,  Arthur Wolf

IPC分类号： G10L21/0208 ,  G10K11/00 ,  G10L21/02 ,  G10L19/02

CPC分类号： G10L21/0208 ,  G10K11/002 ,  G10L19/0204 ,  G10L21/02 ,  G10L2021/02082

摘要： Embodiments of the present invention exploit redundancy of succeeding FFT spectra and use this redundancy for computing interpolated temporal supporting points. An analysis filter bank converts overlapped sequences of an audio (ex. loudspeaker) signal from a time domain to a frequency domain to obtain a time series of short-time loudspeaker spectra. An interpolator temporally interpolates this time series. The interpolation is fed to an echo canceller, which computes an estimated echo spectrum. A microphone analysis filter bank converts overlapped sequences of an audio microphone signal from the time domain to the frequency domain to obtain a time series of short-time microphone spectra. The estimated echo spectrum is subtracted from the microphone spectrum. Further signal enhancement (filtration) may be applied. A synthesis filter bank converts the filtered microphone spectra to the time domain to generate an echo compensated audio microphone signal. Computational complexity of signal processing systems can, therefore, be reduced.