公开(公告)号：US20130325458A1

公开(公告)日：2013-12-05

申请号：US13990176

申请日：2010-11-29

申请人： Markus Buck ,  Timo Matheja ,  Achim Eichentopf

发明人： Markus Buck ,  Timo Matheja ,  Achim Eichentopf

IPC分类号： G10L21/0208

CPC分类号： G10L21/0208 ,  H03G3/3005 ,  H04R3/005 ,  H04R2430/01 ,  H04R2430/03

摘要： A system and method of signal combining that supports different speakers in a noisy environment is provided. Particularly for deviations in the noise characteristics among the channels, various embodiments ensure a smooth transition of the background noise at speaker changes. A modified noise reduction (NR) may achieve equivalent background noise characteristics for all channels by applying a dynamic, channel specific, and frequency dependent maximum attenuation. The reference characteristics for adjusting the background noise may be specified by the dominant speaker channel. In various embodiments, an automatic gain control (AGC) with a dynamic target level may ensure similar speech signal levels in all channels.

摘要翻译： 提供了在嘈杂环境中支持不同扬声器的信号组合系统和方法。 特别是对于通道之间的噪声特性的偏差，各种实施例确保扬声器变化时背景噪声的平滑过渡。 修改的噪声降低（NR）可以通过应用动态，信道特定和频率相关的最大衰减来实现所有信道的等效背景噪声特性。 用于调整背景噪声的参考特性可以由主扬声器通道指定。 在各种实施例中，具有动态目标电平的自动增益控制（AGC）可以确保所有信道中的类似语音信号电平。

公开(公告)号：US20100150375A1

公开(公告)日：2010-06-17

申请号：US12636432

申请日：2009-12-11

申请人： Markus Buck ,  Timo Matheja

发明人： Markus Buck ,  Timo Matheja

IPC分类号： H04B15/00

CPC分类号： G10L25/78 ,  G10L2021/02165

摘要： Embodiments of the invention disclose computer-implemented methods, systems, and computer program products for estimating signal coherence. First, a sound generated by a sound source is detected by a first microphone to obtain a first microphone signal and by a second microphone to obtain a second microphone signal. The first microphone signal is filtered by a first adaptive finite impulse response filter to obtain a first filtered signal. The second microphone signal is filtered by a second adaptive finite impulse response filter, to obtain a second filtered signal. The coherence of the first filtered signal and the second filtered signal is determined based upon the filtered signals. The first and the second microphone signals are filtered such that the difference between the acoustic transfer function for the transfer of the sound from the sound source to the first microphone and the transfer of the sound from the sound source to the second microphone is compensated in the first and second filtered signals.

摘要翻译： 本发明的实施例公开了用于估计信号一致性的计算机实现的方法，系统和计算机程序产品。 首先，由第一麦克风检测由声源产生的声音以获得第一麦克风信号，并由第二麦克风检测第二麦克风信号。 第一麦克风信号被第一自适应有限脉冲响应滤波器滤波以获得第一滤波信号。 第二麦克风信号被第二自适应有限脉冲响应滤波器滤波，以获得第二滤波信号。 基于经滤波的信号确定第一滤波信号和第二滤波信号的相干性。 第一麦克风信号和第二麦克风信号被滤波，以便在声音传输到第一麦克风的声音传递功能与声音从声源传输到第二麦克风之间的差异被补偿在 第一和第二滤波信号。

公开(公告)号：US09888316B2

公开(公告)日：2018-02-06

申请号：US14778643

申请日：2013-03-21

申请人： NUANCE COMMUNICATIONS, INC. ,  Timo Matheja ,  Markus Buck

发明人： Timo Matheja ,  Markus Buck

IPC分类号： H04R3/00 ,  G06F3/16 ,  H04M1/60 ,  H04M1/03

CPC分类号： H04R3/00 ,  G06F3/165 ,  H04M1/03 ,  H04M1/6008 ,  H04R3/005 ,  H04R2430/03 ,  H04R2430/23 ,  H04R2499/11

摘要： Embodiments disclosed herein may include determining a signal parameter of a first microphone and a second microphone associated with a computing device. Embodiments may include generating a reference parameter based upon at least one of the parameter of the first microphone and the parameter of the second microphone. Embodiments may include adjusting a tolerance of at least one of the first microphone and the second microphone, based upon the reference parameter. Embodiments may include receiving, at the first microphone, a first speech signal, the first speech signal having a first speech signal magnitude and receiving, at the second microphone, a second speech signal, the second speech signal having a second speech signal magnitude. Embodiments may include comparing at least one of the first speech signal magnitude and the second speech signal magnitude with a third speech signal magnitude and detecting an obstructed microphone based upon the comparison.

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：US20160050488A1

公开(公告)日：2016-02-18

申请号：US14778643

申请日：2013-03-21

申请人： Timo MATHEJA ,  Markus BUCK ,  NUANCE COMMUNICATIONS, INC.

发明人： Timo Matheja ,  Markus Buck

IPC分类号： H04R3/00

CPC分类号： H04R3/00 ,  G06F3/165 ,  H04M1/03 ,  H04M1/6008 ,  H04R3/005 ,  H04R2430/03 ,  H04R2430/23 ,  H04R2499/11

摘要： Embodiments disclosed herein may include determining a signal parameter of a first microphone and a second microphone associated with a computing device. Embodiments may include generating a reference parameter based upon at least one of the parameter of the first microphone and the parameter of the second microphone. Embodiments may include adjusting a tolerance of at least one of the first microphone and the second microphone, based upon the reference parameter. Embodiments may include receiving, at the first microphone, a first speech signal, the first speech signal having a first speech signal magnitude and receiving, at the second microphone, a second speech signal, the second speech signal having a second speech signal magnitude. Embodiments may include comparing at least one of the first speech signal magnitude and the second speech signal magnitude with a third speech signal magnitude and detecting an obstructed microphone based upon the comparison.

摘要翻译： 本文公开的实施例可以包括确定与计算设备相关联的第一麦克风和第二麦克风的信号参数。 实施例可以包括基于第一麦克风的参数和第二麦克风的参数中的至少一个来生成参考参数。 实施例可以包括基于参考参数调整第一麦克风和第二麦克风中的至少一个的容差。 实施例可以包括在第一麦克风处接收第一语音信号，第一语音信号具有第一语音信号幅度，并且在第二麦克风处接收第二语音信号，第二语音信号具有第二语音信号幅度。 实施例可以包括将第一语音信号幅度和第二语音信号幅度中的至少一个与第三语音信号幅度进行比较，并且基于该比较来检测阻塞的麦克风。

公开(公告)号：US08238575B2

公开(公告)日：2012-08-07

申请号：US12636432

申请日：2009-12-11

申请人： Markus Buck ,  Timo Matheja

发明人： Markus Buck ,  Timo Matheja

IPC分类号： H04B15/00

CPC分类号： G10L25/78 ,  G10L2021/02165

摘要： Embodiments of the invention disclose computer-implemented methods, systems, and computer program products for estimating signal coherence. First, a sound generated by a sound source is detected by a first microphone to obtain a first microphone signal and by a second microphone to obtain a second microphone signal. The first microphone signal is filtered by a first adaptive finite impulse response filter to obtain a first filtered signal. The second microphone signal is filtered by a second adaptive finite impulse response filter, to obtain a second filtered signal. The coherence of the first filtered signal and the second filtered signal is determined based upon the filtered signals. The first and the second microphone signals are filtered such that the difference between the acoustic transfer function for the transfer of the sound from the sound source to the first microphone and the transfer of the sound from the sound source to the second microphone is compensated in the first and second filtered signals.

摘要翻译： 本发明的实施例公开了用于估计信号一致性的计算机实现的方法，系统和计算机程序产品。 首先，由第一麦克风检测由声源产生的声音以获得第一麦克风信号，并由第二麦克风检测第二麦克风信号。 第一麦克风信号被第一自适应有限脉冲响应滤波器滤波以获得第一滤波信号。 第二麦克风信号被第二自适应有限脉冲响应滤波器滤波，以获得第二滤波信号。 基于经滤波的信号确定第一滤波信号和第二滤波信号的相干性。 第一麦克风信号和第二麦克风信号被滤波，以便在声音传输到第一麦克风的声音传递功能与声音从声源传输到第二麦克风之间的差异被补偿在 第一和第二滤波信号。

公开(公告)号：US10326867B2

公开(公告)日：2019-06-18

申请号：US13518404

申请日：2010-02-05

申请人： Tim Haulick ,  Guido Kolano ,  Martin Roessler ,  Hans-Joerg Koepf ,  Walter Schnug ,  Markus Buck

发明人： Tim Haulick ,  Guido Kolano ,  Martin Roessler ,  Hans-Joerg Koepf ,  Walter Schnug ,  Markus Buck

IPC分类号： H04M1/24

摘要： An arrangement is described for measuring performance characteristics of a hands free telephone system. There is a measurement system which is coupleable over a telephone audio interface directly to the hands free telephone system for measuring the performance characteristics.

公开(公告)号：US09580028B2

公开(公告)日：2017-02-28

申请号：US12241837

申请日：2008-09-30

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

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

IPC分类号： G10L15/22 ,  B60R16/037 ,  G10L15/26 ,  G10L17/00

CPC分类号： B60R16/0373 ,  G10L15/22 ,  G10L15/265 ,  G10L17/005

摘要： A speech processing device includes an automotive device that filters data that is sent and received across an in-vehicle bus. The device selectively acquires vehicle data related to a user settings or adjustments. An interface acquires the selected vehicle data from in-vehicle sensors in response to a user's articulation of a first code phrase. A memory stores the selected vehicle data with unique identifying data associated with the user and establishes a connection between the selected vehicle data and the user when a second code phrase is articulated. A data interface provides access to the selected vehicle data and stored relationship data and enables the processing of the data to customize the in-vehicle system. The data interface is responsive to the user's articulation of a third code phrase to process the selected vehicle data that enables the setting or adjustment of the in-vehicle system.

摘要翻译： 语音处理装置包括：汽车装置，其对通过车载总线发送和接收的数据进行滤波。 该设备选择性地获取与用户设置或调整相关的车辆数据。 响应于用户对第一代码短语的表达，接口从车载传感器中获取所选择的车辆数据。 存储器使用与用户相关联的唯一识别数据存储所选择的车辆数据，并且当第二代码短语被铰接时，建立所选择的车辆数据与用户之间的连接。 数据接口提供对所选择的车辆数据和存储的关系数据的访问，并使数据的处理能够定制车载系统。 数据接口响应于用户对第三代码短语的表达，以处理能够设置或调整车载系统的所选车辆数据。

公开(公告)号：US08688458B2

公开(公告)日：2014-04-01

申请号：US11362286

申请日：2006-02-23

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

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

IPC分类号： G10L15/22

CPC分类号： B60R16/0373 ,  G10L2021/02165

摘要： An approach for adjusting an adjustable element, such as a mirror, head rest, steering wheel, heating/air condition blower, associated with a vehicle by determining the position of a speaker in the vehicle.

摘要翻译： 一种用于通过确定车辆中的扬声器的位置来调节与车辆相关联的可调节元件（例如镜子，头枕，方向盘，加热/空调鼓风机）的方法。

公开(公告)号：US08238574B2

公开(公告)日：2012-08-07

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： 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。