ROBUST ADAPTIVE BEAMFORMING WITH ENHANCED NOISE SUPPRESSION
    2.
    发明申请
    ROBUST ADAPTIVE BEAMFORMING WITH ENHANCED NOISE SUPPRESSION 有权
    具有增强噪声抑制功能的稳健自适应光束

    公开(公告)号:US20110274291A1

    公开(公告)日:2011-11-10

    申请号:US13187618

    申请日:2011-07-21

    IPC分类号: H04B15/00

    摘要: A novel adaptive beamforming technique with enhanced noise suppression capability. The technique incorporates the sound-source presence probability into an adaptive blocking matrix. In one embodiment the sound-source presence probability is estimated based on the instantaneous direction of arrival of the input signals and voice activity detection. The technique guarantees robustness to steering vector errors without imposing ad hoc constraints on the adaptive filter coefficients. It can provide good suppression performance for both directional interference signals as well as isotropic ambient noise.

    摘要翻译: 一种具有增强噪声抑制能力的新型自适应波束成形技术。 该技术将声源存在概率纳入自适应阻塞矩阵。 在一个实施例中,基于输入信号的瞬时到达方向和语音活动检测来估计声源存在概率。 该技术保证对导向矢量误差的鲁棒性,而不会对自适应滤波器系数施加自组织约束。 它可以为双向干扰信号以及各向同性环境噪声提供良好的抑制性能。

    TELEPHONY SERVICE INTERACTION MANAGEMENT
    3.
    发明申请
    TELEPHONY SERVICE INTERACTION MANAGEMENT 有权
    电话服务交互管理

    公开(公告)号:US20110238414A1

    公开(公告)日:2011-09-29

    申请号:US12749333

    申请日:2010-03-29

    IPC分类号: G10L19/00

    摘要: A method for managing an interaction of a calling party to a communication partner is provided. The method includes automatically determining if the communication partner expects DTMF input. The method also includes translating speech input to one or more DTMF tones and communicating the one or more DTMF tones to the communication partner, if the communication partner expects DTMF input.

    摘要翻译: 提供了一种用于管理主叫方与通信对方的交互的方法。 该方法包括自动确定通信伙伴是否期望DTMF输入。 如果通信伙伴期望DTMF输入,该方法还包括将语音输入转换成一个或多个DTMF​​音调并将一个或多个DTMF​​音通信给通信对方。

    Systems and methods for real-time audio-visual communication and data collaboration in a network conference environment
    4.
    发明授权
    Systems and methods for real-time audio-visual communication and data collaboration in a network conference environment 有权
    网络会议环境中实时视听通信和数据协作的系统和方法

    公开(公告)号:US07634533B2

    公开(公告)日:2009-12-15

    申请号:US10836778

    申请日:2004-04-30

    IPC分类号: G06F15/16

    摘要: Systems and methods are disclosed that facilitate real-time information exchange in a multimedia conferencing environment. Data Client(s) facilitate data collaboration between users and are maintained separately from audio/video (AV) Clients that provide real-time communication functionality. Data Clients can be remotely located with respect to one another and with respect to a server. A remote user Stand-in Device can be provided that comprises a display to present a remote user to local users, a digital automatic pan/tilt/zoom camera to capture imagery in, for example, a conference room and provide real-time information to an AV Client in a remote office, and a microphone array that can similarly provide real-time audio information from the conference room to an AV Client in the remote office. The invention further facilitates file transfer and presentation broadcast between Data Clients in a single location or in a plurality of disparate locations.

    摘要翻译: 公开了促进多媒体会议环境中的实时信息交换的系统和方法。 数据客户端促进用户之间的数据协作,并与提供实时通信功能的音频/视频(AV)客户端分开维护。 数据客户端可以相对于彼此和相对于服务器远程定位。 可以提供远程用户待机设备,其包括向本地用户呈现远程用户的显示器,用于在例如会议室中捕获图像的数字自动摇摄/俯仰/变焦相机,并且提供实时信息 远程办公室中的AV客户端以及可以类似地从会议室向远程办公室中的AV客户端提供实时音频信息的麦克风阵列。 本发明进一步便于在单个位置或多个不同位置的数据客户端之间的文件传送和呈现广播。

    Conveying Locations In Spoken Dialog Systems
    5.
    发明申请
    Conveying Locations In Spoken Dialog Systems 有权
    输入口语对话系统中的位置

    公开(公告)号:US20090043497A1

    公开(公告)日:2009-02-12

    申请号:US11836955

    申请日:2007-08-10

    IPC分类号: G01C21/34

    CPC分类号: G01C21/3644 G01C21/3679

    摘要: The presentation of location information to a user that is distracted by traveling can result in the user quickly forgetting, or never even comprehending, key parts of the location information, such as the street number. Identification can be made of intersections and points of interest near the user's destination, which can then be provided instead of, or in addition to, the address, thereby increasing user comprehension and retention, especially when distracted. Map data can be parsed into addresses, intersections and points of interest databases. These databases can be accessed to identify proximate intersections and points of interest, which can then be filtered and subsequently ranked to identify one intersection, one point of interest, or both, that can be presented to the user to aid the user in comprehending and retaining the location information even when distracted.

    摘要翻译: 通过旅行分散给用户的位置信息的呈现可能导致用户快速地忘记甚至不理解诸如街道号码的位置信息的关键部分。 识别可以由用户目的地附近的交叉点和兴趣点组成,然后可以提供地址,也可以除了地址之外,还可以提供用户的理解和保留,特别是在分心时。 地图数据可以解析为地址,交叉点和兴趣点数据库。 可以访问这些数据库以识别最近的交叉点和兴趣点,然后可以对这些数据进行过滤并随后进行排序以识别一个交点,一个兴趣点或二者,可以呈现给用户以帮助用户理解和保留 位置信息即使分心。

    Analog preamplifier measurement for a microphone array
    6.
    发明授权
    Analog preamplifier measurement for a microphone array 有权
    用于麦克风阵列的模拟前置放大器测量

    公开(公告)号:US07428309B2

    公开(公告)日:2008-09-23

    申请号:US10772528

    申请日:2004-02-04

    IPC分类号: H04R29/00 H04R3/00

    CPC分类号: H04R3/005

    摘要: An analog preamplifier measurement system for a microphone array builds on conventional microphone arrays by providing an integral “self-calibration system.” This self-calibration system automatically injects an excitation pulse of a known magnitude and phase to all preamplifier inputs within the microphone array. The resulting analog waveform from each preamplifier output is then measured. A frequency analysis, such as, for example, a Fourier or Fast Fourier Transform (FFT), or other conventional frequency analysis, of each of the resulting waveforms is then performed. The results of this frequency analysis are then used to automatically compute frequency-domain compensation gains (e.g., magnitude and phase gains) for each preamplifier for matching or balancing the responses of all of the preamplifiers with each other.

    摘要翻译: 用于麦克风阵列的模拟前置放大器测量系统通过提供集成的“自校准系统”建立在传统麦克风阵列上。 该自校准系统自动将已知幅度和相位的激励脉冲注入到麦克风阵列内的所有前置放大器输入端。 然后测量每个前置放大器输出的结果模拟波形。 然后执行每个所得到的波形的频率分析,例如傅里叶或快速傅里叶变换(FFT)或其他常规频率分析。 然后,该频率分析的结果用于自动计算每个前置放大器的频域补偿增益(例如,幅度和相位增益),用于使所有前置放大器的响应彼此匹配或平衡。

    Spatial noise suppression for a microphone array
    7.
    发明申请
    Spatial noise suppression for a microphone array 有权
    麦克风阵列的空间噪声抑制

    公开(公告)号:US20070150268A1

    公开(公告)日:2007-06-28

    申请号:US11316002

    申请日:2005-12-22

    IPC分类号: G10L21/02

    摘要: A microphone array having at least three microphones provides a captured signal. Spatial noise suppression estimates a desired signal from a captured signal using spatio-temporal distribution of the speech and the noise. In particular, spatial information indicative of at least two quantities of direction are used. A first quantity is based on a first combination of the signals from the at least three microphones, a second quantity is based on a second combination of the signals of the at least three microphones.

    摘要翻译: 具有至少三个麦克风的麦克风阵列提供捕获的信号。 空间噪声抑制使用语音和噪声的时空分布从捕获的信号估计期望的信号。 特别地,使用指示至少两个方向量的空间信息。 第一数量是基于来自至少三个麦克风的信号的第一组合,第二数量是基于至少三个麦克风的信号的第二组合。

    Precision of localization estimates
    8.
    发明申请
    Precision of localization estimates 有权
    本地化估算精度

    公开(公告)号:US20050283328A1

    公开(公告)日:2005-12-22

    申请号:US11210160

    申请日:2005-08-22

    申请人: Ivan Tashev

    发明人: Ivan Tashev

    CPC分类号: G01S7/00

    摘要: Precision and reliability of localization estimates derived from conventional localization systems are improved through a system and method for post-processing of initial localization data, even in environments which may include noise, reflections, or other interference. Such localization systems include conventional sound source localization (SSL) systems based on microphone array inputs, radio source location systems based on directional antenna array inputs, etc. In general, this post-processing system and method applies statistical real-time clustering to initial localization estimates, and then uses this real-time clustering in a multi-stage process to generate new localization estimates having improved precision and reliability relative to the initial localization estimates.

    摘要翻译: 通过用于对初始定位数据进行后处理的系统和方法,即使在可能包括噪声,反射或其他干扰的环境中,通过传统定位系统得到的定位估计的精度和可靠性得到了改进。 这种定位系统包括基于麦克风阵列输入的传统声源定位(SSL)系统,基于定向天线阵列输入的无线电源定位系统等。一般来说,该后处理系统和方法将统计实时聚类应用于初始定位 估计,然后在多阶段过程中使用这种实时聚类,以产生相对于初始定位估计具有改进的精度和可靠性的新的定位估计。

    Localization of mobile computing devices in indoor environments
    9.
    发明授权
    Localization of mobile computing devices in indoor environments 有权
    移动计算设备在室内环境中的本地化

    公开(公告)号:US08548494B2

    公开(公告)日:2013-10-01

    申请号:US13215230

    申请日:2011-08-23

    IPC分类号: H04W24/00

    摘要: Various technologies pertaining to localizing multiple mobile computing devices in an indoor environment are described. Pairs of microphone arrays are selectively positioned in an indoor environment. A localization service assigns a frequency and schedule to a mobile telephone, and the mobile telephone begins outputting vibrations at the assigned frequency and in conformance with the assigned schedule. The microphone arrays sense the vibrations, and angles between the microphone arrays, respectively, and the mobile computing device are computed based upon the sensed vibrations. Such angles are subsequently employed to compute the location of the mobile computing device in the indoor environment.

    摘要翻译: 描述了关于在室内环境中定位多个移动计算设备的各种技术。 一对麦克风阵列选择性地定位在室内环境中。 本地化服务将频率和时间表分配给移动电话,并且移动电话开始以分配的频率输出振动,并且符合分配的时间表。 麦克风阵列感测振动,并且基于感测的振动来计算麦克风阵列和移动计算设备之间的角度。 随后采用这样的角度来计算移动计算设备在室内环境中的位置。

    Adaptive ambient sound suppression and speech tracking
    10.
    发明授权
    Adaptive ambient sound suppression and speech tracking 有权
    自适应环境声音抑制和语音跟踪

    公开(公告)号:US08219394B2

    公开(公告)日:2012-07-10

    申请号:US12690827

    申请日:2010-01-20

    IPC分类号: G10L11/00 G10L21/02 G10L21/00

    摘要: A device for suppressing ambient sounds from speech received by a microphone array is provided. One embodiment of the device comprises a microphone array, a processor, an analog-to-digital converter, and memory comprising instructions stored therein that are executable by the processor. The instructions stored in the memory are configured to receive a plurality of digital sound signals, each digital sound signal based on an analog sound signal originating at the microphone array, receive a multi-channel speaker signal, generate a monophonic approximation signal of the multi-channel speaker signal, apply a linear acoustic echo canceller to suppress a first ambient sound portion of each digital sound signal, generate a combined directionally-adaptive sound signal from a combination of each digital sound signal by a combination of time-invariant and adaptive beamforming techniques, and apply one or more nonlinear noise suppression techniques to suppress a second ambient sound portion of the combined directionally-adaptive sound signal.

    摘要翻译: 提供了一种用于抑制由麦克风阵列接收的语音的环境声音的装置。 该设备的一个实施例包括麦克风阵列,处理器,模数转换器和包含可由处理器执行的存储在其中的指令的存储器。 存储在存储器中的指令被配置为接收多个数字声音信号,基于源自麦克风阵列的模拟声音信号的每个数字声音信号接收多声道扬声器信号,产生多声道扬声器信号的单声道近似信号, 应用线性声学回声消除器来抑制每个数字声音信号的第一环境声音部分,通过时不变和自适应波束成形技术的组合从每个数字声音信号的组合产生组合的定向自适应声音信号 并且应用一个或多个非线性噪声抑制技术来抑制组合的定向自适应声音信号的第二环境声音部分。