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公开(公告)号：US5404397A
公开(公告)日：1995-04-04
申请号：US41802
申请日：1993-04-01
摘要： A conference system comprises speaker units (12) coupled to a central unit (14). The speaker unit (12) comprises an echo canceller (20) with an adaptive filter (38) having an impulse response of comparatively short duration. In a speech pause of the user of the speaker unit (12) the common listening signal (LS) of the conference system is applied to the loudspeaker (18). Of the sounds then picked up by the microphone (16) only those sounds are compensated which are produced by the loudspeaker (18) and which reach the microphone (16) directly and which reach the microphone (16) indirectly via reflections from nearby objects. These sounds have a comparatively short impulse response time and enable a comparatively simple adaptive filter (38) to be used. The other sounds picked up by the microphone (16) reach the microphone (16) via reflections from the walls of the conference space (10) in which the conference system is situated. The other sounds have a comparatively long impulse response time. For the purpose of automatic speech detection these other sounds do not require echo cancellation because they are the same for all the speaker units. Automatic speech detection is now accomplished if the signal strength of the compensated microphone signal (MCCS) produced by the echo canceller (20) exceeds the average signal level of the speaker units.
摘要翻译： 会议系统包括耦合到中央单元（14）的扬声器单元（12）。 扬声器单元（12）包括具有具有相对较短持续时间的脉冲响应的自适应滤波器（38）的回波消除器（20）。 在扬声器单元（12）的用户的语音暂停中，会议系统的公共收听信号（LS）被应用于扬声器（18）。 然后由麦克风拾取的声音（16）中，只有由扬声器（18）产生并直接到达麦克风（16）并通过来自附近物体的反射间接地到达麦克风（16）的那些声音被补偿。 这些声音具有相当短的脉冲响应时间，并且能够使用相对简单的自适应滤波器（38）。 由麦克风拾取的其他声音通过来自会议系统所在的会议空间（10）的墙壁的反射到达麦克风（16）。 其他声音具有比较长的脉冲响应时间。 为了自动语音检测，这些其他声音不需要回声消除，因为它们对于所有扬声器单元是相同的。 如果由回波消除器（20）产生的经补偿的麦克风信号（MCCS）的信号强度超过扬声器单元的平均信号电平，则现在完成自动语音检测。
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2.发明授权Method for storing a parcelwise divided digital data base as well as of addressing a data parcel in a mass memory, and apparatus for carrying out the method 失效用于存储包裹分割数字数据库以及在大容量存储器中寻址数据包的方法，以及用于执行该方法的装置
公开(公告)号：US4888698A
公开(公告)日：1989-12-19
申请号：US110303
申请日：1987-10-19
CPC分类号： G06F17/30955 , G01C21/32 , G06F17/30241 , G06F17/30277 , G11B27/105 , G11B27/28 , G11B2220/20 , G11B2220/213 , G11B2220/2545 , G11B2220/255 , Y10S707/99945 , Y10S707/99948 , Y10S707/99953
摘要： A database is stored in a mass memory. For this purpose, it is first divided into main cells and then into base cells according to a predetermined regular division pattern. Each base cell is then checked to see whether its data content is sufficient to occupy substantially completely a storage parcel having a predetermined capacity. If this is the case, the base cell is thus accommodated in a storage parcel; if this is not the case, adjacent base cells are grouped until a storage parcel is occupied substantially completely. The operation of addressing a storage parcel is effected by the use of a main cell table in which address pointers are stored, each of which points to a base cell table. In the base cell table, an index is given for each base cell and this index indicates in which storage parcel the relevant base cell is accommodated. Each of these indices indicates a location in a data paracel list at which an address indicator is present, which indicates the location at which the relevant parcel is stored in the mass memory.
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3.发明授权Adaptive noise cancelling arrangement, a noise reduction system and a transceiver 失效自适应噪声消除装置，降噪系统和收发器
公开(公告)号：US5740256A
公开(公告)日：1998-04-14
申请号：US762682
申请日：1996-12-11
IPC分类号： G10K11/178 , H04B7/015 , H04B15/00 , H04R3/00 , H04R29/00
CPC分类号： H04B7/015 , H04R3/005 , H04R2430/20 , H04R29/005
摘要： A known cross-coupled adaptive noise cancelling arrangement uses an adaptive noise filter and an adaptive cross-talk filter in a feedback loop for cancelling correlated noise at a primary signal input and reference input. This known cross-coupled ANC does not operate satisfactorily, particularly not for acoustic noise cancellation. This leads to reverberant-like sound signals, in particular in a typical office room with remote noise sources. A different configuration of a cross-coupled adaptive noise cancelling arrangement gives rise to a better performance. The adaptive cross-talk filter is now split into a pre-filter section and an adaptive filter section, the sections using different input signals. The pre-filter section estimates the desired signal from the input signal of the noise cancelling arrangement, and the adaptive filter section has its input coupled to the output of the noise cancelling arrangement, a delay section being provided between the input and the output of the noise cancelling arrangement. In an embodiment, the pre-filter section and the adaptive filter section are separate filters.
摘要翻译： 已知的交叉耦合自适应噪声消除装置在反馈环路中使用自适应噪声滤波器和自适应串扰滤波器来消除主信号输入和参考输入处的相关噪声。 这种已知的交叉耦合ANC不能令人满意地运行，特别是不用于声学噪声消除。 这导致类似混响的声音信号，特别是在具有远程噪声源的典型办公室中。 交叉耦合自适应噪声消除装置的不同配置产生更好的性能。 自适应串扰滤波器现在分为预滤波器部分和自适应滤波器部分，这些部分使用不同的输入信号。 预滤波器部分从噪声消除装置的输入信号估计期望的信号，并且自适应滤波器部分的输入耦合到噪声消除装置的输出，延迟部分设置在输入和输出之间 降噪布置。 在一个实施例中，预滤波器部分和自适应滤波器部分是分离的滤波器。
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公开(公告)号：US07146012B1
公开(公告)日：2006-12-05
申请号：US09196064
申请日：1998-11-19
申请人： Harm J. W. Belt , Cornelis P. Janse
发明人： Harm J. W. Belt , Cornelis P. Janse
IPC分类号： G06F155/00
CPC分类号： G10K15/00 , G10L21/02 , G10L2021/02082 , G10L2021/02166 , H03H17/02 , H04R3/00 , H04R3/02
摘要： In an audio processing arrangement (2) input signals from a plurality of input sources (4,6) are weighted by weighting factors x and y using weighting elements (10, 12). The weighted input signals are combined to a combined signal by an adder (18). The output signal of the adder (18) constitutes the output of the audio processing arrangement.In order to pronounce the signal with the strongest signal, the weighting coefficients (x,y) are controlled to maximize the output signal of the adder (18) under the constraint that the sum of the squares of the weighting coefficients is equal to a constant.
摘要翻译： 在音频处理装置（2）中，使用加权元件（10,12），通过加权因子x和y对来自多个输入源（4,6）的输入信号进行加权。 加权输入信号由加法器（18）组合成组合信号。 加法器（18）的输出信号构成音频处理装置的输出。 为了发出具有最强信号的信号，加权系数（x，y）被控制，以使加权器（18）的输出信号在加权系数的平方和之和等于常数 。
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公开(公告)号：US5390250A
公开(公告)日：1995-02-14
申请号：US810776
申请日：1991-12-17
CPC分类号： H04M9/082
摘要： An echo canceller for use in send/receive apparatus includes a frequency-domain adaptive filter which provides filter parameters to a time-domain programmable filter. The programmable filter produces a signal y(k) which is an estimate of the echo signal produced by acoustic coupling between a loudspeaker in the received path and a microphone in the send path. By subtracting this estimate from the signal produced in the send path, an output signal r(k) is obtained from the send path which is essentially free of echoes. An accurate estimate of the echo signal cannot be obtained if a send signal is present at the input of the send path. Therefore, detecting means in the adaptive filter detects whether such a send signal is present and in that case blocks the transfer of filter parameters from the adaptive filter to the programmable filter. Erroneous modification of the estimated echo signal is thereby avoided. Detection of the presence of a send signal is based on whether the ratio of the power of the signal z(k) in the send path to the power of an output signal of the adaptive filter exceeds a specific threshold.
摘要翻译： 用于发送/接收设备的回波消除器包括频域自适应滤波器，其向时域可编程滤波器提供滤波器参数。 可编程滤波器产生信号y（k），其是通过接收路径中的扬声器与发送路径中的麦克风之间的声耦合产生的回波信号的估计。 通过从在发送路径中产生的信号中减去该估计，从基本上没有回波的发送路径获得输出信号r（k）。 如果发送信号存在于发送路径的输入端，则无法获得回波信号的精确估计。 因此，自适应滤波器中的检测装置检测这样的发送信号是否存在，并且在这种情况下阻止滤波器参数从自适应滤波器到可编程滤波器的传送。 从而避免了估计的回波信号的错误修改。 发送信号的存在的检测基于发送路径中的信号z（k）的功率与自适应滤波器的输出信号的功率的比率是否超过特定阈值。
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公开(公告)号：USRE41445E1
公开(公告)日：2010-07-20
申请号：US10875091
申请日：2004-06-22
申请人： Cornelis P. Janse
发明人： Cornelis P. Janse
IPC分类号： H04M9/08
CPC分类号： H04M9/082
摘要： In an acoustic echocanceller (6), an estimate of an echo signal is determined by an adaptive filter (10) and is subtracted from the input signal by a subtracter (14). The spectrum estimator (12) determines the frequency spectrum of the estimate of the echo signal, and the filter (16) filters the output signal of the subtracter (14) with a filter having a transfer function dependent on the spectrum determined by the estimator (12). The use of this combination results in a substantial improvement of the suppression of the echo signal.
摘要翻译： 在声学回声检测器（6）中，回波信号的估计由自适应滤波器（10）确定，并通过减法器（14）从输入信号中减去。 频谱估计器（12）确定回波信号的估计的频谱，并且滤波器（16）利用具有取决于由估计器确定的频谱的传递函数的滤波器对减法器（14）的输出信号进行滤波 12）。 使用这种组合导致对回波信号抑制的实质性改进。
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公开(公告)号：US07454023B1
公开(公告)日：2008-11-18
申请号：US09310086
申请日：1999-05-11
申请人： Harm J. W. Belt , Cornelis P. Janse
发明人： Harm J. W. Belt , Cornelis P. Janse
CPC分类号： G10K15/00 , G10L21/02 , G10L2021/02082 , G10L2021/02166 , H03H17/02 , H04R3/00 , H04R3/02
摘要： In an audio processing arrangement (2) input signals from a plurality of input sources (4,6) are weighted by weighting factors x and y using weighting elements (10, 12). The weighted input signals are combined to a combined signal by an adder (18). The output signal of the adder (18) constitutes the output of the audio processing arrangement.In order to pronounce the signal with the strongest signal, the weighting coefficients (x,y) are controlled to maximize the output signal of the adder (18) under the constraint that the sum of the squares of the weighting coefficients is equal to a constant.
摘要翻译： 在音频处理装置（2）中，使用加权元件（10,12），通过加权因子x和y对来自多个输入源（4,6）的输入信号进行加权。 加权输入信号由加法器（18）组合成组合信号。 加法器（18）的输出信号构成音频处理装置的输出。 为了发出具有最强信号的信号，加权系数（x，y）被控制，以使加权器（18）的输出信号在加权系数的平方和之和等于常数 。
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8.发明授权Method of storing a topological network, and methods and apparatus for identifying series of 1-cells in a network stored by such a method 失效存储拓扑网络的方法，以及用于识别通过这种方法存储的网络中的1个小区的系列的方法和装置
公开(公告)号：US5754846A
公开(公告)日：1998-05-19
申请号：US769613
申请日：1991-10-01
IPC分类号： G06F17/30
CPC分类号： G06F17/30241 , Y10S707/99942 , Y10S707/99945
摘要： A topological network, comprising a set of 0-cells (nodes) (28, 29) interconnected by a set of 1-cells (aa, bb, . . . , gg).sub.1 is divided into sections (E, F, G) corresponding to discrete parcels of data for storage in a mass memory such as a CD-ROM. A boundary node (28, 29) is defined at each point where the network traverses a boundary between sections. The parcel of data for a given section includes a chain list record for each 1-cell in the section, which chain list record generally refers directly (TP) to a further 1-cell in the network terminating at the same node. However, no chain list record refers directly to a 1-cell outside the section of the network to which the data parcel relates. An indirect reference (0-C') across the section boundary can be made easily, while an overall saving in data volume is obtained compared with a known method. Further information can be introduced into the data parcels by ordering techniques, enabling the network data to be used in a particularly efficient and systematic manner.
摘要翻译： 包括通过一组1-单元（aa，bb，...，gg）1互连的一组0单元（节点）（28,29）的拓扑网络被划分为部分（E，F，G） 对应于用于存储在诸如CD-ROM的大容量存储器中的离散数据包。 边界节点（28,29）被定义在网络遍历区段之间的边界的每个点处。 给定部分的数据包包括该部分中每个1单元的链表记录，该链列表记录通常直接（TP）直接连接到终止于相同节点的网络中的另一个1单元。 然而，没有链表记录直接涉及数据包涉及的网络部分之外的1单元。 与公知的方法相比，可以容易地在部分边界上间接引用（0-C'），同时获得整体的数据量的保存。 可以通过排序技术将更多的信息引入到数据包中，使网络数据以特别有效和系统的方式使用。
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公开(公告)号：US5748751A
公开(公告)日：1998-05-05
申请号：US822958
申请日：1997-03-21
CPC分类号： H04R25/453 , H04R25/502
摘要： In a signal amplifier system, a microphone (2) is connected to an echo canceller (16) via a decorrelator (6). The output signal of the echo canceller (16) is amplified by an amplifier (14) and fed to a loudspeaker (18). The echo canceller (16) is included to avoid instability caused by undesired feedback of the signal coming from the loudspeaker (18) through a feedback path (11). To improve the stabilizing effect of the echo canceller (16), the decorrelator (6) is included for decorrelating the signal coming from the microphone (2) and the signal transmitted by the loudspeaker (18).
摘要翻译： 在信号放大器系统中，麦克风（2）经由去相关器（6）连接到回声消除器（16）。 回波消除器（16）的输出信号由放大器（14）放大并馈送到扬声器（18）。 回波消除器（16）被包括以避免由来自扬声器（18）的信号的不希望的反馈通过反馈路径（11）引起的不稳定性。 为了改善回声消除器（16）的稳定效果，包括去相关器（6）用于对来自麦克风（2）的信号和由扬声器（18）传输的信号进行解相关。
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公开(公告)号：US5610991A
公开(公告)日：1997-03-11
申请号：US350357
申请日：1994-12-06
申请人： Cornelis P. Janse
发明人： Cornelis P. Janse
IPC分类号： G10L15/20 , G10L21/0208 , G10L21/0216 , H04B7/26 , H04R3/00
CPC分类号： G10L21/0208 , G10L2021/02166
摘要： A noise reduction system and device, and a mobile radio station. Known is a combined Zelinski-spectral subtraction system (1) for noise reduction in a combined speech signal (a(t)) in which signals are recorded with a plurality of microphones (5, 6, 7), using a Wiener filter (10) for estimation of the combined speech signal (a(t)'). In the known system (1) sums and differences of all combinations of speech signals are formed, it being assumed that the differences comprise noise only. Furthermore, a two stage estimation process is carried out, giving rise to considerable estimation errors. An alternative combined Zelinski-spectral subtraction system (1) is proposed, giving rise to fewer estimation errors and being more efficient from a computational point of view. In the Zelinski system, spectral subtraction is carried out on a combined cross spectrum (.PHI..sub.cc). Then, on a speech segment by speech segment basis, filter coeffients for the Wiener filter (10) are determined from a combined auto power spectrum (.PHI..sub.ac) and the thus corrected combined cross power spectrum (.PHI..sub.cc '). The spectral subtraction is carried out on a lower part of the frequency range only, thereby not introducing unneccesary artefacts.
摘要翻译： 降噪系统和设备，以及移动无线电台。 已知的是使用维纳滤波器（10（10））在信号与多个麦克风（5,6,7）一起记录的组合语音信号（a（t））中降噪的组合Zelinski谱减法系统（1） ）用于估计组合语音信号（a（t）'）。 在已知的系统（1）中，形成了语音信号的所有组合的和差，假定差异仅包括噪声。 此外，进行两级估计处理，导致相当大的估计误差。 提出了一种替代的组合Zelinski谱减法系统（1），从计算的角度来看，估计误差较小，效率更高。 在Zelinski系统中，频谱相减在组合交叉谱（PHI cc）上进行。 然后，在基于语音段的语音段上，根据组合的自动功率谱（PHI ac）和由此校正的组合交叉功率谱（PHI cc'）确定维纳滤波器（10）的滤波系数。 频谱减法仅在频率范围的较低部分进行，从而不引入不必要的伪像。
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