Lucent Technologies Inc.

Agere Systems Inc.

Man Mohan Sondhi

Jacob Benesty

Steven Leslie Gay

Tomas Fritz Gaensler

Gregory Peter Kochanski

Gary Wayne Elko

James Edward West

Dennis Raymond Morgan

Ingrid Birgitta Carlbom

Sarma Vgk Pingali

Thomas A Funkhouser

Joseph Lindley Hall, Ii

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.

Adaptive filter for network echo cancellation

Audio signals from any array of microphones are individually filtered, delayed, and scaled in order to form an acoustic beam that focuses the array on a particular region. Nonlinear robust signal estimation processing is applied to the resulting set of audio signals to generate an output signal for the array. The nonlinear robust signal estimation processing may involve dropping or otherwise reducing the magnitude of one or more of the highest and lowest data in each set of values from the resulting audio signals and then selecting the median from or generating an average of the remaining values to produce a representative, central value for the output audio signal. The nonlinear robust signal estimation processing effectively discriminates against noise originating at an unknown location outside of the focal region of the acoustic beam.

Acoustic beam forming with robust signal estimation

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.

Acoustic modeling system and method using pre-computed data structures for beam tracing and path generation

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.

Dynamic allocation of resources for echo cancellation

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.

Adaptive filter utilizing proportional affine projection algorithm

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.

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Mountain Lakes, NJ, United States of America

Man Mohan Sondhi