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：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：US08204248B2

公开(公告)日：2012-06-19

申请号：US12104836

申请日：2008-04-17

申请人： Tim Haulick ,  Gerhard Uwe Schmidt ,  Markus Buck ,  Tobias Wolff

发明人： Tim Haulick ,  Gerhard Uwe Schmidt ,  Markus Buck ,  Tobias Wolff

IPC分类号： H04R3/00

CPC分类号： H04R3/005 ,  G01S3/00 ,  G01S3/043 ,  G01S3/802 ,  G01S3/808 ,  G01S3/8083 ,  G01S5/0252 ,  G01S5/0273 ,  G01S5/18 ,  G01S5/30 ,  G01S7/292 ,  G01S7/52003 ,  G01S7/523 ,  G01S13/003 ,  G01S13/42 ,  G01S15/00 ,  G01S15/003 ,  G01S15/06 ,  G01S15/42 ,  G01S15/876 ,  G10K11/34 ,  G10K15/08 ,  H04B7/086 ,  H04M1/6075 ,  H04R1/40 ,  H04R1/406 ,  H04R29/00 ,  H04R29/001 ,  H04R2201/401 ,  H04R2201/403 ,  H04R2203/12 ,  H04R2410/01 ,  H04R2430/20 ,  H04R2499/11 ,  H04S7/00 ,  H04S7/30 ,  H04S7/301 ,  H04S7/302 ,  H04S7/303 ,  H04S7/305

摘要： A system locates a speaker in a room containing a loudspeaker and a microphone array. The loudspeaker transmits a sound that is partly reflected by a speaker. The microphone array detects the reflected sound and converts the sound into a microphone signal. A processor determines the speaker's direction relative to the microphone array, the speaker's distance from the microphone array, or both, based on the characteristics of the microphone signals.

摘要翻译： 系统将扬声器定位在包含扬声器和麦克风阵列的房间中。 扬声器传输部分由扬声器反射的声音。 麦克风阵列检测反射的声音，并将声音转换成麦克风信号。 基于麦克风信号的特性，处理器确定扬声器相对于麦克风阵列的方向，扬声器距离麦克风阵列或两者的距离。

公开(公告)号：US20120106749A1

公开(公告)日：2012-05-03

申请号：US13273816

申请日：2011-10-14

申请人： Markus Buck ,  Tim Haulick

发明人： Markus Buck ,  Tim Haulick

IPC分类号： H04B3/20 ,  H04R3/00

CPC分类号： H04R3/005 ,  H04R2430/25

摘要： A microphone compensation system responds to changes in the characteristics of individual microphones in an array of microphones. The microphone compensation system provides a communication system with consistent performance despite microphone aging, widely varying environmental conditions, and other factors that alter the characteristics of the microphones. Furthermore, lengthy, complex, and costly measurement and analysis phases for determining initial settings for filters in the communication system are eliminated.

摘要翻译： 麦克风补偿系统可以响应麦克风阵列中各个麦克风的特性变化。 麦克风补偿系统提供了具有一致性能的通信系统，尽管麦克风老化，广泛变化的环境条件以及改变麦克风特性的其它因素。 此外，消除了用于确定通信系统中的滤波器的初始设置的冗长，复杂且昂贵的测量和分析阶段。

公开(公告)号：US08098848B2

公开(公告)日：2012-01-17

申请号：US11749678

申请日：2007-05-16

申请人： Tim Haulick ,  Gerhard Uwe Schmidt ,  Markus Buck

发明人： Tim Haulick ,  Gerhard Uwe Schmidt ,  Markus Buck

IPC分类号： H03G5/00

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：US07020291B2

公开(公告)日：2006-03-28

申请号：US10121180

申请日：2002-04-12

申请人： Markus Buck ,  Tim Haulick ,  Klaus Linhard

发明人： Markus Buck ,  Tim Haulick ,  Klaus Linhard

IPC分类号： H04R3/00

CPC分类号： G10L21/0208 ,  G10L2021/02165

摘要： The present invention relates to a method with which speech is captured in a noisy environment with as high a speech quality as possible. To this end, a compact array of, for example, two single microphones is combined to form one system through signal processing methods consisting of adaptive beam formation and spectral subtraction. Through the combination with a spectral subtraction, the reference signal of the beam former is freed from speech signal components to the extent that a reference signal of the interference is formed and the beam former produces high gains.

公开(公告)号：US09620146B2

公开(公告)日：2017-04-11

申请号：US14398769

申请日：2012-05-16

申请人： Markus Buck ,  Tim Haulick ,  Timo Matheja

发明人： Markus Buck ,  Tim Haulick ,  Timo Matheja

IPC分类号： G10L15/00 ,  G10L25/48 ,  H04M3/56 ,  H04R3/00 ,  H04R3/12 ,  H04R27/00 ,  H04M1/60

CPC分类号： G10L21/0205 ,  G10L15/00 ,  G10L15/32 ,  G10L21/0272 ,  G10L25/48 ,  G10L2021/02166 ,  H04M1/6075 ,  H04M3/568 ,  H04R3/005 ,  H04R3/12 ,  H04R27/00 ,  H04R2227/009 ,  H04R2499/13

摘要： A multi-mode speech communication system is described that has different operating modes for different speech applications. A speech service compartment contains multiple system users, multiple input microphones that develop microphone input signals from the system users to the system, and multiple output loudspeakers that develop loudspeaker output signals from the system to the system users. A signal processing module is in communication with the speech applications and includes an input processing module and an output processing module. The input processing module processes the microphone input signals to produce a set user input signals for each speech application that are limited to currently active system users for that speech application. The output processing module processes application output communications from the speech applications to produce loudspeaker output signals to the system users, wherein for each different speech application, the loudspeaker output signals are directed only to system users currently active in that speech application. The signal processing module dynamically controls the processing of the microphone input signals and the loudspeaker output signals to respond to changes in currently active system users for each application.

公开(公告)号：US08483406B2

公开(公告)日：2013-07-09

申请号：US12843632

申请日：2010-07-26

申请人： Markus Buck ,  Tim Haulick

发明人： Markus Buck ,  Tim Haulick

IPC分类号： H04R3/00

CPC分类号： H04R29/005 ,  G10L21/0216 ,  G10L21/0232 ,  G10L2021/02166 ,  H04R3/005 ,  H04R2410/07 ,  H04R2430/20

摘要： A system for detecting noise in a signal received by a microphone array and a method for detecting noise in a signal received by a microphone array is disclosed. The system also provides for the reduction of noise in a signal received by a microphone array and a method for reducing noise in a signal received by a microphone array. The signal to noise ratio in handsfree systems may be improved, particularly in handsfree systems present in a vehicular environment.

公开(公告)号：US08275145B2

公开(公告)日：2012-09-25

申请号：US11740164

申请日：2007-04-25

申请人： Markus Buck ,  Tim Haulick ,  Gerhard Uwe Schmidt

发明人： Markus Buck ,  Tim Haulick ,  Gerhard Uwe Schmidt

IPC分类号： H04B1/00 ,  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分量与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：US08194872B2

公开(公告)日：2012-06-05

申请号：US11234837

申请日：2005-09-23

申请人： Markus Buck ,  Tim Haulick ,  Phillip A. Hetherington ,  Pierre Zakarauskas

发明人： Markus Buck ,  Tim Haulick ,  Phillip A. Hetherington ,  Pierre Zakarauskas

IPC分类号： A61F11/06 ,  H04B15/00 ,  H04R1/40

CPC分类号： G10L21/0208 ,  G10L21/0216 ,  G10L2021/02166 ,  H04R3/005 ,  H04R2430/25

摘要： An adaptive signal processing system eliminates noise from input signals while retaining desired signal content, such as speech. The resulting low noise output signal delivers improved clarity and intelligibility. The low noise output signal also improves the performance of subsequent signal processing systems, including speech recognition systems. An adaptive beamformer in the signal processing system consistently updates beamforming signal weights in response to changing microphone signal conditions. The adaptive weights emphasize the contribution of high energy microphone signals to the beamformed output signal. In addition, adaptive noise cancellation logic removes residual noise from the beamformed output signal based on a noise estimate derived from the microphone input signals.

摘要翻译： 自适应信号处理系统消除来自输入信号的噪声，同时保留期望的信号内容，例如语音。 所产生的低噪声输出信号提供了更好的清晰度和清晰度。 低噪声输出信号还提高了后续信号处理系统的性能，包括语音识别系统。 信号处理系统中的自适应波束形成器响应于变化的麦克风信号状况持续更新波束成形信号权重。 自适应权重强调高能麦克风信号对波束形成的输出信号的贡献。 此外，自适应噪声消除逻辑基于从麦克风输入信号导出的噪声估计，从波束形成的输出信号中去除残余噪声。