Huawei Technologies Co., Limited

Wenyu Jin

Willem Bastiaan Kleijn

Yue Lang

Peter Grosche

Richard Heusdens

Thomas Sherson

David Virette

Panji Setiawan

Simone Fontana

Kainan Chen

Wei Xiao

A device estimates direction of arrival (DOA) of sound fromsound sources received by P microphones, wherein P&#x2265;&#x3e;1. The device is configured to transform the output signals of the microphones into the frequency domain and compute a covariance matrix for each of N frequency bins in a range of frequencies of the sound. Further, the device is configured to calculate an adapted covariance matrix from each of the covariance matrices for wide-band merging, calculate an accumulated covariance matrix from the N adapted covariance matrices, and estimate the DOA for each of the sound sources based on the accumulated covariance matrix. In order to calculate an adapted covariance matrix from a covariance matrix, the device is configured to spectrally decompose the covariance matrix and obtain a plurality of eigenvectors, rotate each obtained eigenvector, and construct each rotated eigenvector back to the shape of the covariance matrix.

Device and method for estimating direction of arrival of sound from a plurality of sound sources

A sound processing node is provided for an arrangement of sound processing nodes configured to receive a plurality of sound signals. The sound processing node comprises a processor configured to generate an output signal based on the plurality of sound signals weighted by a plurality of beamforming weights. The processor is configured to adaptively determine the plurality of beamforming weights on the basis of an adaptive linearly constrained minimum variance beamformer using a transformed version of a least mean squares formulation of a constrained gradient descent approach. The transformed version of the least mean squares formulation of the constrained gradient descent approach is based on a transformation of the least mean squares formulation of the constrained gradient descent approach to the dual domain.

Sound processing node of an arrangement of sound processing nodes

A signal generator has a filter bank that provides weighted versions of audio signals to speakers. The weights were derived by identifying a first constraint that limits a weight that can be applied to an audio signal to be provided to a first speaker. A characteristic of a second speaker that affects how a user will perceive audio signals output by that speaker relative to audio signals output by the first speaker was also determined. A second constraint was determined based on the determined characteristic and the first constraint. The weights were then determined so as to minimize a difference between an actual balance of each signal that is expected to be heard by a user and a target balance. The signal generator can achieve sweet spot correction and sound stage widening simultaneously. It also achieves a balanced sound stage, particularly when the speakers are asymmetric.

Apparatus and method for weighting stereo audio signals

An apparatus for processing soundfield data is provided. The soundfield data defines a soundfield within a spatial reproduction region comprising at least one bright zone and at least one quiet zone. The apparatus comprises an applicator configured to apply a spatially continuously varying weighting function to the soundfield data in order to obtain weighted soundfield data defining a weighted soundfield, wherein the spatially continuously varying weighting function is configured to enhance the soundfield in at least one of the bright zone and the quiet zone.

Apparatus and method for processing soundfield data

A sound field device is disclosed that comprises an elevation cue estimator, a low-frequency filter estimator, and a high-frequency filter estimator. The elevation cue-estimator is configured to estimate an elevation cue of a head-related transfer function (HRTF) of at least one listener. The low-frequency filter estimator is configured to estimate one or more low-frequency filter elements based on the elevation cue. The high-frequency filter estimator is configured to estimate one or more high-frequency filter elements based on the elevation cue. An estimation method of the low-frequency filter estimator is different from an estimation method of the high-frequency filter estimator. The one or more low-frequency filter elements and the one or more high-frequency filter elements are for driving an array of loudspeakers to generate an elevated sound impression at a bright zone.

Method and device for generating an elevated sound impression

A sound processing node for an arrangement of sound processing nodes is disclosed. The sound processing nodes being configured to receive a plurality of sound signals, wherein the sound processing node comprises a processor configured to determine a beamforming signal on the basis of the plurality of sound signals weighted by a plurality of weights, wherein the processor is configured to determine the plurality of weights using a transformed version of a linearly constrained minimum variance approach, the transformed version of the linearly constrained minimum variance approach being obtained by applying a convex relaxation to the linearly constrained minimum variance approach.

A signal processor for determining a plurality of drive signals for driving a plurality of loudspeakers to cancel a reverberation effect in a listening area, wherein the signal processor is configured to determine from one or more measured audio signals a plurality of measured physical coefficients in a basis of physical sound functions, such that a sum of the physical sound functions, weighted with the plurality of measured physical coefficients approximates the one or more measured audio signals, wherein at least half of the plurality of measured physical coefficients are zero, determine a residual error between the plurality of measured physical coefficients and a plurality of desired physical coefficients, estimate a transfer function describing a transformation from the plurality of desired physical coefficients to the plurality of measured physical coefficients, based on the determined residual error, and update the plurality of drive signals based on the estimated transfer function.

Adaptive reverberation cancellation system

An audio rendering system is provided that comprises a plurality of loudspeakers arranged to approximate a desired spatial sound field within a predetermined reproduction region, wherein the loudspeakers are configured to approximate the sound field based on a weighted series of orthonormal basis functions for the reproduction region.

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Munich, Germany

Wenyu Jin