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公开(公告)号：US20080267378A1
公开(公告)日：2008-10-30
申请号：US11569170
申请日：2005-05-20
IPC分类号： H04M3/42
摘要： A conference system (1) comprises a central unit (2) and speaker units (3) which are connectable to the central unit. The central unit (2), which serves to combine speech signals from the speaker units (3) and to distribute the combined speech signals to said units, comprises an adaptive filter (23) for suppressing feedback. Each speaker unit (3) comprises a microphone (33), a loudspeaker (34), an activation switch (35) and an adaptive filter (36) coupled between the microphone (33) and the loudspeaker (34). When the speaker unit is not activated, the adaptive filter (36) serves as an echo canceller, while serving as a feedback suppressor when the speaker unit is activated. By keeping the loudspeaker (34) always on, any transients due to mis-adaptations of the filter (36) are avoided.
摘要翻译： 会议系统（1）包括可连接到中央单元的中央单元（2）和扬声器单元（3）。 用于组合来自扬声器单元（3）的语音信号并将组合语音信号分配给所述单元的中央单元（2）包括用于抑制反馈的自适应滤波器（23）。 每个扬声器单元（3）包括麦克风（33），扬声器（34），激活开关（35）和耦合在麦克风（33）和扬声器（34）之间的自适应滤波器（36）。 当扬声器单元未被激活时，自适应滤波器（36）用作回声消除器，同时当扬声器单元被激活时用作反馈抑制器。 通过保持扬声器（34）始终处于打开状态，避免了由于过滤器（36）的错误适应引起的任何瞬变。
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公开(公告)号：US20110007073A1
公开(公告)日：2011-01-13
申请号：US12920132
申请日：2009-03-03
摘要： A digital image (110) which is displayed to a user (118) is modified to include an aspect (120) of a detected at least one characteristic of the user (118) to give the user (118) the impression that the user (118) is present within the scene displayed by the digital image (110).
摘要翻译： 对用户（118）显示的数字图像（110）被修改为包括用户（118）的检测到的至少一个特征的方面（120），以向用户（118）给出用户（118） 118）存在于由数字图像（110）显示的场景内。
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公开(公告)号：US20100189274A1
公开(公告)日：2010-07-29
申请号：US12664467
申请日：2008-06-16
IPC分类号： H04B3/20
CPC分类号： G10L21/02 , G10L2021/02082 , H04M9/082
摘要： A method is provided which is suitable to cope with non-linear echo paths during acoustic echo cancellation in speakerphones. Non-linear paths occur particularly in hands-free operation of, e.g., a mobile phone, due to driving the amplifier and loudspeaker in the non-linear range. The idea is to combine the commonly known one microphone approach of linear acoustic echo cancellation using an adaptive filter and a post-processor together with a multiple microphone approach using beam forming which separately removes the non-linear part of the echo.
摘要翻译： 提供一种适用于在扬声器电话中的声学回声消除期间处理非线性回声路径的方法。 由于在非线性范围内驱动放大器和扬声器，所以非线性路径特别发生在例如移动电话的免提操作中。 这个想法是将使用自适应滤波器和后处理器的通常已知的一种麦克风方法与使用波束形成的多麦克风方法相结合，分离地去除回波的非线性部分。
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公开(公告)号：US20100020984A1
公开(公告)日：2010-01-28
申请号：US12513529
申请日：2007-11-08
IPC分类号： H04B15/00
CPC分类号： H04R3/02 , H04R2499/13
摘要： A signal processing system comprises a microphone (20), a subtractor (22) arranged to receive an output of the microphone (20), an amplifier G arranged to receive an output of the subtractor (22), a rear loudspeaker (24) arranged to receive an output of the amplifier G, a front loudspeaker (26) arranged to receive an output of the amplifier G, and one or more summers (28) interposed between the amplifier G and a loudspeaker (24, 26), the or each summer (28) arranged to add an audio signal m[k] to the signal s[k] received from the amplifier G. The system also includes a mixing matrix D arranged to receive the respective inputs R, F of the rear and front loudspeakers (24, 26) and arranged to output a summation signal R+F and a difference signal R−F, and an adaptive filter SAF; MCAF arranged to receive the outputs R+F, R−F of the mixing matrix D, the subtractor (22) arranged to receive an output of the adaptive filter SAF; MCAF and an output of the subtractor (22) arranged to control the adaptive filter SAF; MCAF.
摘要翻译： 信号处理系统包括麦克风（20），布置成接收麦克风（20）的输出的减法器（22），布置成接收减法器（22）的输出的放大器G，布置成 以接收放大器G的输出，布置成接收放大器G的输出的前置扬声器（26）和插在放大器G和扬声器（24,26）之间的一个或多个加法器（28），所述或每个 夏季（28）被布置为向从放大器G接收的信号s [k]添加音频信号m [k]。该系统还包括混合矩阵D，其被布置成接收后置和前置扬声器的相应输入R，F （24,26），并且被配置为输出求和信号R + F和差分信号RF，以及自适应滤波器SAF; MCAF被布置为接收混合矩阵D的输出R + F，R-F，减法器（22）被布置为接收自适应滤波器SAF的输出; MCAF和用于控制自适应滤波器SAF的减法器（22）的输出; MCAF。
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5.发明授权Sound reinforcement system having an echo suppressor and loudspeaker beamformer 失效具有回波抑制器和扬声器波束形成器的扩声系统公开(公告)号：US07054451B2
公开(公告)日：2006-05-30
申请号：US10483854
申请日：2002-06-24
IPC分类号： H04R27/00
CPC分类号： H04R3/02 , H04R3/005 , H04R3/12 , H04R27/00 , H04R2201/403
摘要： A sound reinforcement system includes several microphones, a microphone beamformer coupled to the microphones, adaptive echo compensator (EC) coupled to the microphone beamformer for generating an echo compensated microphone signal, and several loudspeakers coupled to the adaptive EC. An adaptive loudspeaker beamformer is coupled between the adaptive EC and the loudspeakers for shaping the directional pattern of the loudspeakers. The adaptive loudspeaker beamformer creates a beam pattern which is capable of creating a “null” in the direction of speaker(s) such that howling is effectively prevented.
摘要翻译： 扩音系统包括几个麦克风，耦合到麦克风的麦克风波束形成器，耦合到麦克风波束形成器的自适应回波补偿器（EC），用于产生回波补偿的麦克风信号，以及耦合到自适应EC的若干扬声器。 自适应扬声器波束形成器耦合在自适应EC和扬声器之间，用于对扬声器的方向图进行整形。 自适应扬声器波束形成器产生能够在扬声器的方向上产生“零”的波束图案，从而有效地防止啸声。
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公开(公告)号：US08781818B2
公开(公告)日：2014-07-15
申请号：US13141710
申请日：2009-12-17
申请人： Cornelis Pieter Janse , Leon C. A. Van Stuivenberg , Harm Jan Willem Belt , Bahaa Eddine Sarroukh , Mahdi Triki
发明人： Cornelis Pieter Janse , Leon C. A. Van Stuivenberg , Harm Jan Willem Belt , Bahaa Eddine Sarroukh , Mahdi Triki
IPC分类号： G10L19/00
CPC分类号： H04R3/005 , H04M1/6033 , H04M3/56 , H04R2430/20 , H04S7/303 , H04S2400/11
摘要： The invention proposes extracting one or more speech signals (151-154) as well as one or more ambient signals (131) from sound signals captured by microphones, wherein each of the speech signals corresponds to a different speaker. The invention proposes to transmit both the one or more speech signals (151-154) and the one or more ambient signals (131) to a rendering side, as opposed to sending only speech signals. This enables to reproduce the speech and ambient signals in a spatially different way at the rendering side. By reproducing the ambient signals a feeling of “being together” is created. In an embodiment, the invention enables reproducing two or more speech signals spatially from each other and from the ambient signals so that speech intelligibility is increased despite the presence of the ambient signals.
摘要翻译： 本发明提出从由麦克风捕获的声音信号中提取一个或多个语音信号（151-154）以及一个或多个环境信号（131），其中每个语音信号对应于不同的扬声器。 本发明提出将一个或多个语音信号（151-154）和一个或多个环境信号（131）发送到呈现侧，而不是仅发送语音信号。 这使得能够在渲染侧以空间不同的方式再现语音和环境信号。 通过再现环境信号，产生“在一起”的感觉。 在一个实施例中，本发明使得能够在空间上从环境信号中再现两个或更多个语音信号，使得尽管存在环境信号，语音可懂度也增加。
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公开(公告)号：US08655003B2
公开(公告)日：2014-02-18
申请号：US13322636
申请日：2010-05-27
CPC分类号： H04R1/1041 , H04R1/1083 , H04R5/033 , H04R25/456 , H04R25/652 , H04R2420/07 , H04R2460/01 , H04R2460/05 , H04R2460/11
摘要： An earphone arrangement comprises a microphone (109) which generates a microphone signal and a sound transducer (101) which radiates a first sound component to a user's ear (103) in response to a drive signal. An acoustic channel (111) is further provided for channeling external sound so as to provide a second sound component to the user's ear (103). An acoustic valve (117) allows the attenuation of the acoustic channel (111) to be controlled in response to a valve control signal. A control circuit (105) generates the valve control signal in response to the microphone signal to provide a variable attenuation resulting in a mixed sound of the first sound component and the second sound component reaching the user's ear (103). The combined use of acoustic and e.g. electric signal paths allows improved performance and in particular allows a dynamic trade-off between open and closed earphone design characteristics with respect to external sounds.
摘要翻译： 耳机装置包括产生麦克风信号的麦克风（109）和响应于驱动信号向用户的耳朵（103）辐射第一声音分量的声音换能器（101）。 进一步提供声通道（111）用于引导外部声音，以向用户的耳朵（103）提供第二声音分量。 声门（117）允许响应于阀控制信号来控制声通道（111）的衰减。 控制电路（105）响应于麦克风信号产生气门控制信号，以提供可变衰减，导致第一声音分量和第二声音分量的混合声音到达用户的耳朵（103）。 组合使用声学和例如。 电信号路径允许改进的性能，并且特别地允许在开放和闭合的耳机设计特征之间相对于外部声音的动态权衡。
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公开(公告)号：US08403105B2
公开(公告)日：2013-03-26
申请号：US13133839
申请日：2009-12-11
CPC分类号： G01S5/22 , G01S3/8034 , G10L2021/02165 , H04N7/142 , H04N7/147 , H04N7/15 , H04S7/30 , H04S2400/11
摘要： A sound source location is estimated by particle filtering where a set of particles represents a probability density function for a state variable comprising the sound source location. The method includes determining the weight for a particle in response to a correlation between estimated acoustic transfer functions from the sound source to at least two sound recording positions. A weight update function may specifically be determined deterministically from the correlation and thus the correlation may be used as a pseudo-likelihood function for the measurement function of the particle filtering. The acoustic transfer functions may be determined from an audio beamforming towards the sound source. The audio weight may be combined with a video weight to generate a multi-modal particle filtering approach.
摘要翻译： 通过粒子滤波来估计声源位置，其中一组粒子表示包括声源位置的状态变量的概率密度函数。 该方法包括响应于从声源到至少两个录音位置的估计的声学传递函数之间的相关性来确定颗粒的权重。 权重更新功能可以从相关性确定地确定地确定，因此可以将相关性用作用于粒子滤波的测量函数的伪似然函数。 声传递函数可以从对声源形成的音频波束确定。 音频权重可以与视频权重组合以产生多模式粒子滤波方法。
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公开(公告)号：US20120082335A1
公开(公告)日：2012-04-05
申请号：US13322636
申请日：2010-05-27
IPC分类号： H04R1/10
CPC分类号： H04R1/1041 , H04R1/1083 , H04R5/033 , H04R25/456 , H04R25/652 , H04R2420/07 , H04R2460/01 , H04R2460/05 , H04R2460/11
摘要： An earphone arrangement comprises a microphone (109) which generates a microphone signal and a sound transducer (101) which radiates a first sound component to a user's ear (103) in response to a drive signal. An acoustic channel (111) is further provided for channeling external sound so as to provide a second sound component to the user's ear (103). An acoustic valve (117) allows the attenuation of the acoustic channel (111) to be controlled in response to a valve control signal. A control circuit (105) generates the valve control signal in response to the microphone signal to provide a variable attenuation resulting in a mixed sound of the first sound component and the second sound component reaching the user's ear (103). The combined use of acoustic and e.g. electric signal paths allows improved performance and in particular allows a dynamic trade-off between open and closed earphone design characteristics with respect to external sounds.
摘要翻译： 耳机装置包括产生麦克风信号的麦克风（109）和响应于驱动信号向用户的耳朵（103）辐射第一声音分量的声音换能器（101）。 进一步提供声通道（111）用于引导外部声音，以向用户的耳朵（103）提供第二声音分量。 声门（117）允许响应于阀控制信号来控制声通道（111）的衰减。 控制电路（105）响应于麦克风信号产生气门控制信号，以提供可变衰减，导致第一声音分量和第二声音分量的混合声音到达用户的耳朵（103）。 组合使用声学和例如。 电信号路径允许改进的性能，并且特别地允许在开放和闭合的耳机设计特征之间相对于外部声音的动态权衡。
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10.发明授权Adaptive beamformer, sidelobe canceller, handsfree speech communication device 有权自适应波束形成器，旁瓣消除器，免提语音通信设备公开(公告)号：US07957542B2
公开(公告)日：2011-06-07
申请号：US11568240
申请日：2005-04-20
IPC分类号： H04R3/00
CPC分类号： G10K11/341
摘要： The adaptive beamformer unit (191) comprises: a filtered sum beamformer (107) arranged to process input audio signals (u 1, u2) from an array of respective microphones (101, 103), and arranged to yield as an output a first audio signal (z) predominantly corresponding to sound from a desired audio source (160) by filtering with a first adaptive filter (f1(-t)) a first one of the input audio signals (u1) and with a second adaptive filter (f2(-t)) a second one of the input audio signals (u2), the coefficients of the first filter (f1(-t)) and the second filter (f2(-t)) being adaptable with a first step size (a1) and a second step size ((x2) respectively; noise measure derivation means (111) arranged to derive from the input audio signals (u1, u2) a first noise measure (x1) and a second noise measure (x2); and an updating unit (192) arranged to determine the first and second step size (a1, (x2) with an equation comprising in a denominator the first noise measure (x1) for the first step size (a1), respectively the second noise measure (x2) for the second step size (a2). This makes the beamformer relatively robust against the influence of correlated audio interference. The beamformer may also be incorporated in a sidelobe canceller topology yielding a more noise cleaned desired sound estimate, which can be used in a related, more advanced adaptive filter (f1(-t), f2(-t)) updating. Such a beamformer is typically useful for application in handsfree speech communication systems.
摘要翻译： 自适应波束形成器单元（191）包括：经滤波的和波束形成器（107），被布置成从相应麦克风（101,103）的阵列处理输入音频信号（u 1，u 2），并且被布置为产生作为输出的第一音频 信号（z）主要对应于来自所需音频源（160）的声音，通过用输入音频信号（u1）中的第一个和第二自适应滤波器（f2（-t））的第一自适应滤波器（f1（-t））进行滤波） -t））输入音频信号（u2）中的第二个，第一滤波器（f1（-t））和第二滤波器（f2（-t））的系数可适应第一步长（a1） 和第二步长（（x2）），噪声测量导出装置（111）被布置成从输入音频信号（u1，u2）导出第一噪声测量（x1）和第二噪声测量（x2）;以及更新 单元（192），其布置成以分母包括第一步长的第一噪声测量（x1）来确定第一和第二步长（a1，（x2）） （a1）分别为第二步长（a2）的第二噪声测量（x2）。 这使得波束形成器相对于相关音频干扰的影响相对较强。 波束形成器还可以并入在旁瓣消除器拓扑中，产生更多噪声清除的期望声音估计，其可以在相关的更高级的自适应滤波器（f1（-t），f2（-t））更新中使用。 这种波束形成器通常用于免提语音通信系统中的应用。
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