公开(公告)号：US06751322B1

公开(公告)日：2004-06-15

申请号：US09165312

申请日：1998-10-02

申请人： Ingrid B. Carlbom ,  Gary W. Elko ,  Thomas A. Funkhouser ,  Sarma V. Pingali ,  Man Mohan Sondhi ,  James Edward West

发明人： Ingrid B. Carlbom ,  Gary W. Elko ,  Thomas A. Funkhouser ,  Sarma V. Pingali ,  Man Mohan Sondhi ,  James Edward West

IPC分类号： H03G300

CPC分类号： H04S3/00

摘要： A system and method for acoustic modeling partitions an input 3D spatial model into convex cells, and constructs a cell adjacency data structure representing the neighbor relationships between adjacent cells. For each sound source located in the spatial environment, convex pyramidal beams are traced through the input spatial model via recursive depth-first traversal of the cell-adjacency graph. During beam tracing, a beam tree data structure is constructed to encode propagation paths, which may include specular reflection, transmission, diffuse reflection, and diffraction events, from the source location to regions of the input spatial model. The beam tree data structure is then accessed for real-time computation and auralization of propagation paths to an arbitrary receiver location.

摘要翻译： 用于声学建模的系统和方法将输入3D空间模型分割成凸小区，并且构建表示相邻小区之间的邻居关系的小区邻接数据结构。 对于位于空间环境中的每个声源，凸金字塔束通过输入空间模型通过细胞邻接图的递归深度优先遍历来跟踪。 在光束跟踪期间，构造波束树数据结构以对从输入空间模型的源位置到区域的传播路径进行编码，其可以包括镜面反射，透射，漫反射和衍射事件。 然后访问光束树数据结构，用于实时计算和传播路径到任意接收器位置的声音。

公开(公告)号：US07072465B1

公开(公告)日：2006-07-04

申请号：US09228772

申请日：1999-01-06

申请人： Jacob Benesty ,  Tomas Fritz Gaensler ,  Steven Leslie Gay ,  Man Mohan Sondhi

发明人： Jacob Benesty ,  Tomas Fritz Gaensler ,  Steven Leslie Gay ,  Man Mohan Sondhi

IPC分类号： H04B3/23

CPC分类号： H04B3/234 ,  H03H2021/0063

摘要： A robust adaptive filter for use in a network echo canceller or other digital signal processing application utilizes a coefficient vector update device that, through the application of fast converging algorithms to a fast impulse response filter yields fast convergence of the adaptive filter's characteristics with the avoidance of divergence due to the onset of double talk. Robustness is also provided, via an adaptive scale non-linearity device which applies an adaptive scale non-linearity to the filter algorithms fed to the fast impulse response filter by the coefficient vector update device, so that the samples of an echo signal to be cancelled which are taken during the onset of double talk can be handled in such a manner that after the double talk detector causes adaptation to cease, the initial, potentially disturbing samples do not cause significant divergence in the filter system.

摘要翻译： 用于网络回波消除器或其他数字信号处理应用的鲁棒自适应滤波器利用系数向量更新装置，其通过将快速收敛算法应用于快速脉冲响应滤波器产生自适应滤波器特征的快速收敛，避免了 由于双重谈话的开始，分歧。 还通过自适应标度非线性装置提供鲁棒性，其对由系数向量更新装置馈送到快速脉冲响应滤波器的滤波器算法施加自适应标度非线性，使得要消除的回波信号的样本 可以以双方通话检测器引起适应停止的方式处理在双重通话开始期间拍摄的，初始的潜在干扰的样本不会在滤波器系统中引起明显的发散。

公开(公告)号：US5828756A

公开(公告)日：1998-10-27

申请号：US747730

申请日：1996-11-12

申请人： Jacob Benesty ,  Joseph Lindley Hall, II ,  Dennis Raymond Morgan ,  Man Mohan Sondhi

发明人： Jacob Benesty ,  Joseph Lindley Hall, II ,  Dennis Raymond Morgan ,  Man Mohan Sondhi

IPC分类号： H04R3/02 ,  H04B3/23 ,  H04M3/18 ,  H04M3/56 ,  H04M9/08 ,  H04B3/20

CPC分类号： H04M9/082

摘要： A method and apparatus for estimating individual impulse responses for a stereophonic communication system, such as a teleconferencing system, which involves selectively reducing the correlation between the individual channel signals of the stereophonic system. Selective reduction of stereophonic source signal correlation advantageously results in the estimation of individual impulse responses of a receiving room of the stereophonic communication system. The selectively reduced-correlation source signals are provided to conventional adaptive filters and the receiving room loudspeakers. Automatic echo cancellation is performed in a conventional fashion, but on the selectively reduced-correlation source signals. Specifically, selective reduction of source signal correlation between two stereophonic channels of a teleconferencing system is achieved by introducing a small non-linearity into each channel in order to reduce the interchannel coherence. In accordance with certain illustrative embodiments of the present invention, each channel signal has added thereto a non-linear function of the channel signal itself, thereby reducing the interchannel coherence while preserving the quality of the signal. In one particular embodiment, the non-linear function comprises the half-wave rectifier.

摘要翻译： 一种用于估计诸如电话会议系统的立体声通信系统的各个脉冲响应的方法和装置，其涉及选择性地降低立体声系统的各个声道信号之间的相关性。 立体声源信号相关性的选择性降低有利地导致对立体声通信系统的接收室的各个脉冲响应的估计。 选择性的相关源信号被提供给传统的自适应滤波器和接收室扬声器。 以常规方式执行自动回波消除，但是在选择性地减小相关源信号上。 具体来说，电话会议系统的两个立体声道之间的源信号相关性的选择性降低是通过在每个通道中引入小的非线性来实现的，以便减少通道间的相干性。 根据本发明的某些说明性实施例，每个信道信号已经添加了信道信号本身的非线性函数，从而在保持信号质量的同时减少了信道间相干性。 在一个特定实施例中，非线性功能包括半波整流器。

公开(公告)号：US06751314B1

公开(公告)日：2004-06-15

申请号：US09431847

申请日：1999-11-02

申请人： Jacob Benesty ,  Tomas Fritz Gaensler ,  Steven Leslie Gay ,  Man Mohan Sondhi

发明人： Jacob Benesty ,  Tomas Fritz Gaensler ,  Steven Leslie Gay ,  Man Mohan Sondhi

IPC分类号： H04M100

CPC分类号： H04B3/23

摘要： A method is presented for dynamic resource allocation in a speech signal echo canceler enabling more efficient echo cancellation and as a result the ability for an Integrated Circuit to handle additional channels than heretofore possible. This is accomplished by applying one or more of three efficiency enhancing strategies. First, no update of coefficients is computed or convolution performed if the power level of the far end speech signal is below a given threshold. Second, convolution is limited to the set of active taps (i.e., taps that account for most of the power in the echo). Third, new coefficients are computed only when the power of the error signal is greater than a given threshold. Lastly, the set of active coefficients is periodically updated. These strategies release computational resources from unnecessary computations and divert them to other channels that may be active.

摘要翻译： 提出了一种用于语音信号回波消除器中的动态资源分配的方法，其实现更有效的回波消除，并且因此能够使集成电路处理比以前更多的附加信道。 这是通过应用三种效率提高策略中的一种或多种实现的。 首先，如果远端语音信号的功率电平低于给定阈值，则不会计算系数更新或卷积。 第二，卷积限于有效抽头的集合（即，回波中大部分功率的抽头）。 第三，只有当误差信号的功率大于给定的阈值时才计算新的系数。 最后，定期更新该有源系数集。 这些策略从不必要的计算中释放计算资源，并将其转移到可能处于活动状态的其他通道。

公开(公告)号：US06744886B1

公开(公告)日：2004-06-01

申请号：US09227327

申请日：1999-01-06

申请人： Jacob Benesty ,  Tomas Fritz Gaensler ,  Steven Leslie Gay ,  Man Mohan Sondhi

发明人： Jacob Benesty ,  Tomas Fritz Gaensler ,  Steven Leslie Gay ,  Man Mohan Sondhi

IPC分类号： H04M908

CPC分类号： H04B3/234 ,  H03H2021/0063

摘要： An adaptive filter suitable for network echo cancellation and other applications contains a coefficient vector update device for feeding coefficient vector updates to a finite impulse response filter in accordance with fast converging algorithms. A double talk detector is included for causing filter adaptation to cease in the presence of double talk in the system being echo cancelled. The coefficient vector update device utilizes a proportional affine projection algorithm to provide fast convergence of the filter system and improved performance over other filter devices utilizing different fast converging algorithms.

摘要翻译： 适用于网络回波消除和其他应用的自适应滤波器包含用于根据快速收敛算法向有限脉冲响应滤波器馈送系数向量更新的系数向量更新装置。 包括双通话检测器，用于在系统中存在双重通话时停止响应消除过滤器适配。 系数向量更新装置利用比例仿射投影算法提供了滤波系统的快速收敛，并且利用不同的快速收敛算法提高了超过其它滤波器的性能。