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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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5.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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公开(公告)号：US20080300869A1
公开(公告)日：2008-12-04
申请号：US11572278
申请日：2005-07-18
CPC分类号： G10L21/02 , G10L2021/02082 , H04B3/23
摘要： A method of estimating the reverberations in an acoustic signal (y) comprises the steps of determining the frequency spectrum (Y) of the signal (y), providing a first parameter (α) indicative of the decay of the reverberations part (r) of the signal over time, and providing a second parameter (β) indicative of the amplitude of the direct part (d) of the signal relative to the reverberations part (r). An estimated frequency spectrum ({hacek over (R)}) of the reverberations signal (r) is produced using the frequency spectrum (Y) of a previous frame, the first parameter (α), and the second parameter (β). The second parameter (β). The second parameter (β) is preferably inversely proportional to the early-to-late ratio of the signal (y).
摘要翻译： 估计声信号（y）中的混响的方法包括确定信号（y）的频谱（Y）的步骤，提供指示混响部分（r）的衰减的第一参数（α） 并且提供指示信号相对于混响部分（r）的直接部分（d）的振幅的第二参数（β）。 使用前一帧的频谱（Y），第一参数（α）和第二参数（β）产生混响信号（r）的估计频谱（{hacek over（R）}）。 第二个参数（beta）。 第二参数（β）优选地与信号（y）的早到晚比成反比。
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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：US20080085009A1
公开(公告)日：2008-04-10
申请号：US11576915
申请日：2005-10-13
CPC分类号： H04M9/082 , G10L2021/02082 , H04B3/23 , H04R3/002
摘要： An echo cancellation device (1) comprises a first adaptive filter (13) for producing a first echo cancellation signal (y1), a second adaptive filter (15) for producing a second echo cancellation signal (y2), and a post-processor (18) for suppressing any remaining echo. The first adaptive filter (13) and the second adaptive filter (15) are designed for canceling a first (e.g. direct) part of the echo impulse response and a second (e.g. diffuse) part of the echo impulse response respectively. The device (1) may be utilized in a mobile telephone.
摘要翻译： 回波消除装置（1）包括用于产生第一回波消除信号（y1> 1）的第一自适应滤波器（13），用于产生第二回波抵消信号（y 2）和用于抑制任何剩余回波的后处理器（18）。 第一自适应滤波器（13）和第二自适应滤波器（15）被设计用于分别消除回波脉冲响应的第一（例如直接）部分和回波脉冲响应的第二（例如扩散）部分。 设备（1）可以用在移动电话中。
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公开(公告)号：US08472655B2
公开(公告)日：2013-06-25
申请号：US12997889
申请日：2009-06-17
CPC分类号： G10L21/0208 , G10L2021/02166 , H04R3/005 , H04R25/407
摘要： An audio processing arrangement (200) comprises a plurality of audio sources (101, 102) generating input audio signals, a processing circuit (110) for deriving processed audio signals from the input audio signals, a combining circuit (120) for deriving a combined audio signal from the processed audio signals, and a control circuit (130) for controlling the processing circuit in order to maximize a power measure of the combined audio signal and for limiting a function of gains of the processed audio signals to a predetermined value. In accordance with the present invention, the audio processing arrangement (200) comprises a pre-processing circuit (140) for deriving pre-processed audio signals from the input audio signals to minimize a cross-correlation of interferences comprised in the input audio signals. The pre-processed signals are provided to the processing circuit (110) instead of the input audio signals.
摘要翻译： 音频处理装置（200）包括产生输入音频信号的多个音频源（101,102），用于从输入音频信号中导出经处理的音频信号的处理电路（110），用于导出组合的组合电路 来自经处理的音频信号的音频信号，以及用于控制处理电路以便最大化组合音频信号的功率测量并且将经处理的音频信号的增益的功能限制到预定值的控制电路（130）。 根据本发明，音频处理装置（200）包括用于从输入音频信号导出预处理音频信号的预处理电路（140），以最小化包含在输入音频信号中的干扰的互相关。 预处理的信号被提供给处理电路（110）而不是输入的音频信号。
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公开(公告)号：US20120063609A1
公开(公告)日：2012-03-15
申请号：US13321995
申请日：2010-05-27
申请人： Mahdi Triki , Cornelis Pieter Janse
发明人： Mahdi Triki , Cornelis Pieter Janse
IPC分类号： H04B3/20
摘要： A multi-channel acoustic echo canceller arrangement comprises a microphone (111) providing a microphone signal having contributions from at least two audio sources (107, 109) to be cancelled. An echo canceling circuit (113, 115) performs echo cancellation of the two audio sources (107, 109) based on channel estimates for channels from each of the audio sources (107, 109) to the microphone (111). An estimation circuit (117) generates each of the channel estimates as a combination of a previous channel estimate and a channel estimate update where the combination includes applying a relative weight to the channel estimate update relative to the previous channel estimate. A weight processor 119 varies the relative weight in response to a time value. The arrangement may provide improved echo-cancellation for scenarios wherein the rendering of sound from the audio sources (107, 109) is time varying, such as when time varying decorrelation filters are used.
摘要翻译： 多通道声学回声消除器装置包括麦克风（111），其提供具有来自至少两个音频源（107,109）的贡献的麦克风信号以被取消。 回声消除电路（113,115）基于从每个音频源（107,109）到麦克风（111）的信道的信道估计，执行两个音频源（107,109）的回波消除。 估计电路（117）产生每个信道估计作为先前信道估计和信道估计更新的组合，其中组合包括相对于先前信道估计对信道估计更新应用相对权重。 加权处理器119响应于时间值改变相对权重。 该布置可以提供改善的回声消除，其中来自音频源（107,109）的声音的渲染是时变的，例如当使用时变解相关滤波器时。
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