Non-linear echo cancellation
    11.
    发明授权
    Non-linear echo cancellation 有权
    非线性回波消除

    公开(公告)号:US09065895B2

    公开(公告)日:2015-06-23

    申请号:US13623468

    申请日:2012-09-20

    Inventor: Jes Thyssen

    CPC classification number: H04M9/082

    Abstract: A two-stage structure for performing non-linear echo cancellation is described in which a first echo canceller is used to attenuate linear echo components of a microphone signal and a second echo canceller is used to attenuate non-linear echo components of the output signal generated by the first echo canceller. One or both of the echo cancellers may be implemented using closed-form solutions, including a closed form solution for a hybrid method in the frequency domain.

    Abstract translation: 描述了用于执行非线性回波消除的两级结构,其中使用第一回波消除器来衰减麦克风信号的线性回波分量,并且使用第二回波消除器来衰减产生的输出信号的非线性回波分量 由第一个回波消除器。 回波消除器中的一个或两个可以使用封闭形式的解决方案来实现,包括用于频域中的混合方法的闭合形式的解决方案。

    NON-LINEAR ECHO CANCELLATION
    12.
    发明申请
    NON-LINEAR ECHO CANCELLATION 有权
    非线性ECHO取消

    公开(公告)号:US20130216056A1

    公开(公告)日:2013-08-22

    申请号:US13623468

    申请日:2012-09-20

    Inventor: Jes Thyssen

    CPC classification number: H04M9/082

    Abstract: A two-stage structure for performing non-linear echo cancellation is described in which a first echo canceller is used to attenuate linear echo components of a microphone signal and a second echo canceller is used to attenuate non-linear echo components of the output signal generated by the first echo canceller. One or both of the echo cancellers may be implemented using closed-form solutions, including a closed form solution for a hybrid method in the frequency domain.

    Abstract translation: 描述了用于执行非线性回波消除的两级结构,其中使用第一回波消除器来衰减麦克风信号的线性回波分量,并且使用第二回波消除器来衰减产生的输出信号的非线性回波分量 由第一个回波消除器。 回波消除器中的一个或两个可以使用封闭形式的解决方案来实现,包括用于频域中的混合方法的闭合形式的解决方案。

    MULTI-MICROPHONE SOURCE TRACKING AND NOISE SUPPRESSION
    13.
    发明申请
    MULTI-MICROPHONE SOURCE TRACKING AND NOISE SUPPRESSION 审中-公开
    多麦克风源跟踪和噪声抑制

    公开(公告)号:US20160241955A1

    公开(公告)日:2016-08-18

    申请号:US15136708

    申请日:2016-04-22

    Abstract: Methods, systems, and apparatuses are described for improved multi-microphone source tracking and noise suppression. In multi-microphone devices and systems, frequency domain acoustic echo cancellation is performed on each microphone input, and microphone levels and sensitivity are normalized. Methods, systems, and apparatuses are also described for improved acoustic scene analysis and source tracking using steered null error transforms, on-line adaptive acoustic scene modeling, and speaker-dependent information. Switched super-directive beamforming reinforces desired audio sources and closed-form blocking matrices suppress desired audio sources based on spatial information derived from microphone pairings. Underlying statistics are tracked and used to updated filters and models. Automatic detection of single-user and multi-user scenarios, and single-channel suppression using spatial information, non-spatial information, and residual echo are also described.

    Abstract translation: 描述了改进的多麦克风源跟踪和噪声抑制的方法,系统和装置。 在多麦克风设备和系统中,在每个麦克风输入端执行频域声学回声消除,并对麦克风电平和灵敏度进行归一化。 还描述了用于改进的声场分析和源跟踪的方法,系统和装置,其使用转向零误差变换,在线自适应声场建模和与扬声器相关的信息。 开关超方向波束成形加强了所需的音频源,并且封闭形式的阻塞矩阵基于从麦克风配对得到的空间信息来抑制所需音频源。 跟踪统计信息,并用于更新过滤器和模型。 还描述了单用户和多用户场景的自动检测,以及使用空间信息,非空间信息和残余回声的单信道抑制。

    SPEAKER-IDENTIFICATION-ASSISTED SPEECH PROCESSING SYSTEMS AND METHODS
    14.
    发明申请
    SPEAKER-IDENTIFICATION-ASSISTED SPEECH PROCESSING SYSTEMS AND METHODS 有权
    语音识别辅助语音处理系统和方法

    公开(公告)号:US20140278417A1

    公开(公告)日:2014-09-18

    申请号:US13965661

    申请日:2013-08-13

    CPC classification number: G10L17/00 G10L17/06 G10L21/00

    Abstract: Methods, systems, and apparatuses are described for performing speaker-identification-assisted speech processing. In accordance with certain embodiments, a communication device includes speaker identification (SID) logic that is configured to identify a user of the communication device and/or the identity of a far-end speaker participating in a voice call with a user of the communication device. Knowledge of the identity of the user and/or far-end speaker is then used to improve the performance of one or more speech processing algorithms implemented on the communication device.

    Abstract translation: 描述了用于执行说话者识别辅助语音处理的方法,系统和装置。 根据某些实施例,通信设备包括说话人标识(SID)逻辑,其被配置为识别通信设备的用户和/或与通信设备的用户参与语音呼叫的远端扬声器的身份 。 然后使用对用户和/或远端扬声器的身份的知识来改善在通信设备上实现的一个或多个语音处理算法的性能。

    Systems and methods for presenting audio messages
    15.
    发明授权
    Systems and methods for presenting audio messages 有权
    用于呈现音频消息的系统和方法

    公开(公告)号:US08781080B2

    公开(公告)日:2014-07-15

    申请号:US13959636

    申请日:2013-08-05

    Abstract: Systems and methods for presenting audio messages are provided. In some aspects, a method includes receiving an audio message from a first user and generating a text-based representation of the audio message. The method also includes generating one or more identification tags based on the text-based representation of the audio message. At least one of the one or more identification tags includes a subject of the audio message. The method also includes presenting at least one of the text-based representation of the audio message or the one or more identification tags to a second user using a graphical user interface.

    Abstract translation: 提供用于呈现音频消息的系统和方法。 在一些方面,一种方法包括从第一用户接收音频消息并生成音频消息的基于文本的表示。 该方法还包括基于音频消息的基于文本的表示来生成一个或多个识别标签。 一个或多个识别标签中的至少一个包括音频消息的对象。 该方法还包括使用图形用户界面向第二用户呈现音频消息或一个或多个识别标签的基于文本的表示中的至少一个。

    SYSTEM AND METHOD FOR LOUDSPEAKER PROTECTION
    16.
    发明申请

    公开(公告)号:US20170345438A1

    公开(公告)日:2017-11-30

    申请号:US15365268

    申请日:2016-11-30

    Inventor: Jes Thyssen

    Abstract: Systems, devices, and methods are described for providing loudspeaker protection and distortion suppression. An upstream loudspeaker model estimation component receives sensed electrical characteristics of a loudspeaker and generates an impedance model from which an excursion model of the loudspeaker and a gain change parameter may be generated. The impedance components are fitted to features of an estimated impedance, based on the sensed characteristics, to generate the estimated impedance model that is converted to an excursion model of the loudspeaker. A downstream audio signal processing component, based on the excursion model, or parameters thereof, limits a predicted excursion of the loudspeaker utilizing excursion-constraining processing circuitry that includes a non-linear constraint filter. Processed audio signals associated with the limited excursion are subject to distortion suppression prior to releasing the output audio signals for playback on the loudspeaker. Distortion suppression is performed based on a determined relationship between processed and pre-processed audio signals.

    Integrated speech intelligibility enhancement system and acoustic echo canceller
    17.
    发明授权
    Integrated speech intelligibility enhancement system and acoustic echo canceller 有权
    集成语音清晰度增强系统和声学回声消除器

    公开(公告)号:US09361901B2

    公开(公告)日:2016-06-07

    申请号:US14145775

    申请日:2013-12-31

    CPC classification number: G10L21/0208 G10L19/012 G10L21/0232

    Abstract: A system and method is described that improves the intelligibility of a far-end telephone speech signal to a user of a telephony device in the presence of near-end background noise. As described herein, the system and method improves the intelligibility of the far-end telephone speech signal in a manner that does not require user input and that minimizes the distortion of the far-end telephone speech signal. The system is integrated with an acoustic echo canceller and shares information therewith.

    Abstract translation: 描述了一种在存在近端背景噪声的情况下提高远端电话语音信号给​​电话设备的用户的可懂度的系统和方法。 如这里所述,该系统和方法以不需要用户输入并使远端电话语音信号的失真最小化的方式提高了远端电话语音信号的可懂度。 该系统与声学回声消除器集成并与其共享信息。

    Echo cancellation using closed-form solutions
    18.
    发明授权
    Echo cancellation using closed-form solutions 有权
    使用闭式解决方案进行回声消除

    公开(公告)号:US09036826B2

    公开(公告)日:2015-05-19

    申请号:US13720672

    申请日:2012-12-19

    CPC classification number: H04M9/082

    Abstract: A system that utilizes closed-form solutions to perform echo cancellation is described. The system includes a filter, filter parameter determination logic and a combiner. The filter is configured to process a far-end audio signal in accordance with one or more filter parameters to generate an estimated echo signal. The filter parameter determination logic is configured to update estimated statistics associated with the far-end audio signal and a microphone signal based on instantaneous statistics associated with the far-end audio signal and the microphone signal, and calculate the one or more filter parameters based upon the updated estimated statistics. The combiner is configured to generate an estimated near-end audio signal by subtracting the estimated echo signal from the microphone signal.

    Abstract translation: 描述了利用闭式解决方案来执行回波消除的系统。 该系统包括滤波器,滤波器参数确定逻辑和组合器。 滤波器被配置为根据一个或多个滤波器参数处理远端音频信号以产生估计回波信号。 滤波器参数确定逻辑被配置为基于与远端音频信号和麦克风信号相关联的瞬时统计信息来更新与远端音频信号相关联的估计统计量和麦克风信号,并且基于以下方式计算一个或多个滤波器参数 更新的估计统计数据。 组合器被配置为通过从麦克风信号中减去估计的回波信号来产生估计的近端音频信号。

    MULTI-MICROPHONE SOURCE TRACKING AND NOISE SUPPRESSION
    19.
    发明申请
    MULTI-MICROPHONE SOURCE TRACKING AND NOISE SUPPRESSION 有权
    多麦克风源跟踪和噪声抑制

    公开(公告)号:US20140286497A1

    公开(公告)日:2014-09-25

    申请号:US14216769

    申请日:2014-03-17

    Abstract: Methods, systems, and apparatuses are described for improved multi-microphone source tracking and noise suppression. In multi-microphone devices and systems, frequency domain acoustic echo cancellation is performed on each microphone input, and microphone levels and sensitivity are normalized. Methods, systems, and apparatuses are also described for improved acoustic scene analysis and source tracking using steered null error transforms, on-line adaptive acoustic scene modeling, and speaker-dependent information. Switched super-directive beamforming reinforces desired audio sources and closed-form blocking matrices suppress desired audio sources based on spatial information derived from microphone pairings. Underlying statistics are tracked and used to updated filters and models. Automatic detection of single-user and multi-user scenarios, and single-channel suppression using spatial information, non-spatial information, and residual echo are also described.

    Abstract translation: 描述了改进的多麦克风源跟踪和噪声抑制的方法,系统和装置。 在多麦克风设备和系统中,在每个麦克风输入端执行频域声学回声消除,并对麦克风电平和灵敏度进行归一化。 还描述了用于改进的声场分析和源跟踪的方法,系统和装置,其使用转向零误差变换,在线自适应声场建模和与扬声器相关的信息。 开关超方向波束成形加强了所需的音频源,并且封闭形式的阻塞矩阵基于从麦克风配对得到的空间信息来抑制所需音频源。 跟踪统计信息,并用于更新过滤器和模型。 还描述了单用户和多用户场景的自动检测,以及使用空间信息,非空间信息和残余回声的单信道抑制。

    INTEGRATED SPEECH INTELLIGIBILITY ENHANCEMENT SYSTEM AND ACOUSTIC ECHO CANCELLER
    20.
    发明申请
    INTEGRATED SPEECH INTELLIGIBILITY ENHANCEMENT SYSTEM AND ACOUSTIC ECHO CANCELLER 有权
    集成语音智能增强系统和耳塞式收音机

    公开(公告)号:US20140188466A1

    公开(公告)日:2014-07-03

    申请号:US14145775

    申请日:2013-12-31

    CPC classification number: G10L21/0208 G10L19/012 G10L21/0232

    Abstract: A system and method is described that improves the intelligibility of a far-end telephone speech signal to a user of a telephony device in the presence of near-end background noise. As described herein, the system and method improves the intelligibility of the far-end telephone speech signal in a manner that does not require user input and that minimizes the distortion of the far-end telephone speech signal. The system is integrated with an acoustic echo canceller and shares information therewith.

    Abstract translation: 描述了一种在存在近端背景噪声的情况下提高远端电话语音信号给​​电话设备的用户的可懂度的系统和方法。 如这里所述,该系统和方法以不需要用户输入并使远端电话语音信号的失真最小化的方式提高了远端电话语音信号的可懂度。 该系统与声学回声消除器集成并与其共享信息。

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