-
公开(公告)号：US09888316B2
公开(公告)日：2018-02-06
申请号：US14778643
申请日：2013-03-21
发明人： Timo Matheja , Markus Buck
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：US20160050488A1
公开(公告)日：2016-02-18
申请号：US14778643
申请日：2013-03-21
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：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：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
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：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：US11367437B2
公开(公告)日：2022-06-21
申请号：US16426356
申请日：2019-05-30
发明人： Timo Matheja , Markus Buck , Andreas Kirbach , Martin Roessler , Tim Haulick , Julien Premont , Josef Anastasiadis , Rudi Vuerinckx , Christophe Ris , Stijn Verschaeren , Hakan Ari , Dieter Ranz
摘要： There is provided a speech dialog system that includes a first microphone, a second microphone, a processor and a memory. The first microphone captures first audio from a first spatial zone, and produces a first audio signal. The second microphone captures second audio from a second spatial zone, and produces a second audio signal. The processor receives the first audio signal and the second audio signal, and the memory contains instructions that control the processor to perform operations of a speech enhancement module, an automatic speech recognition module, and a speech dialog module that performs a zone-dedicated speech dialog.
-
公开(公告)号：US10229686B2
公开(公告)日：2019-03-12
申请号：US15329354
申请日：2014-08-18
发明人： Markus Buck , Tobias Herbig , Simon Graf , Christophe Ris
摘要： Methods and apparatus to process microphone signals by a speech enhancement module to generate an audio stream signal including first and second metadata for use by a speech recognition module. In an embodiment, speech recognition is performed using endpointing information including transitioning from a silence state to a maybe speech state, in which data is buffered, based on the first metadata and transitioning to a speech state, in which speech recognition is performed, based upon the second metadata.
-
9.发明申请公开(公告)号：US20190073999A1
公开(公告)日：2019-03-07
申请号：US16074953
申请日：2016-02-10
发明人： Julien Prémont , Tim Haulick , Emanuele Dalmasso , Munir Nikolai Alexander Georges , Andreas Kellner , Gaetan Martens , Oliver van Porten , Holger Quast , Martin Rößler , Tobias Wolff , Markus Buck
摘要： According to some aspects, a system for detecting a designated wake-up word is provided, the system comprising a plurality of microphones to detect acoustic information from a physical space having a plurality of acoustic zones, at least one processor configured to receive a first acoustic signal representing the acoustic information received by the plurality of microphones, process the first acoustic signal to identify content of the first acoustic signal originating from each of the plurality of acoustic zones, provide a plurality of second acoustic signals, each of the plurality of second acoustic signals substantially corresponding to the content identified as originating from a respective one of the plurality of acoustic zones, and performing automatic speech recognition on each of the plurality of second acoustic signals to determine whether the designated wake-up word was spoken.
-
公开(公告)号：US09978389B2
公开(公告)日：2018-05-22
申请号：US15442843
申请日：2017-02-27
发明人： Markus Buck , Tim Haulick , Timo Matheja
IPC分类号： G10L15/00 , G10L21/02 , G10L21/0272 , G10L21/0216 , G10L15/32
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 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 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.
-
-
-
-
-
-
-
-
-
搜索结果
135 条记录 (耗时 0.029 秒)
