Adaptive reverberation cancellation system

US10199032B2 · US · B2

Patent metadata
FieldValue
Publication numberUS-10199032-B2
Application numberUS-201815952864-A
CountryUS
Kind codeB2
Filing dateApr 13, 2018
Priority dateOct 14, 2015
Publication dateFeb 5, 2019
Grant dateFeb 5, 2019

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Abstract

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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.

First claim

Opening claim text (preview).

The invention claimed is: 1. A sound device comprising: a signal processor 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 a plurality of drive signals based on the estimated transfer function. 2. The sound device of claim 1 , wherein the signal processor is further configured to, when determining the plurality of measured physical coefficients; minimize an error measure between the measured audio signals and a linear transformation of the measured physical coefficients; and minimize a number of non-zero entries of the plurality of measured physical coefficients. 3. The sound device of claim 2 , wherein the signal processor is further configured to, when minimizing the error measure and minimizing the number of non-zero entries of the plurality of measured physical coefficients, determine a vector b of the plurality of measured physical coefficients according to: b =argmin y ∥y∥ p p , such that ∥ v−Φy∥ 2 ≤∈ for 0≤ p≤ 1, wherein ∥y∥ p is a p-norm of a vector y, Φ is a M×N sensing matrix comprising columns with the physical sound functions, N»M, v is an M×1 observation vector which comprises the one or more measured audio signals corresponding to M locations within the listening area, wherein the signal processor is further configured to randomly chose the M locations. 4. The sound device of claim 1 , wherein the basis of physical sound functions is orthogonal with regard to an inner product that for a first vector bi and a second vector bj is representable as: b i |b j =∫ R b i ( x ) b j ( x ) w ( x ) dx=σ ij , wherein R is a reproduction region of a plurality of loudspeakers, w(x) is a weighting function, and σ ij is 1 for i=j and 0 otherwise. 5. The sound device of claim 1 , wherein the basis of physical sound functions comprises an orthonormal set of physical sound functions obtained from a modified Gram-Schmidt process on plane wave functions corresponding to a plurality of angles. 6. The sound device of claim 1 , wherein the transfer function assigns a zero-coupling between a first coefficient and a second coefficient of the basis of physical sound functions, wherein the transfer function is representable as a diagonal matrix U(k). 7. The sound device of claim 6 , wherein the signal processor is further configured to, when estimating the transfer function, estimate the diagonal matrix U(k) using a Least Mean Squares filter and/or using a Recursive Least Squares filter. 8. The sound device of claim 7 , wherein the signal processor is further configured to, when estimating the diagonal matrix U(k), compute an n-th element of the diagonal matrix U(k) according to U ^ n ⁡ ( k ) τ H = U ^ n ⁡ ( k ) τ - 1 H + 1 ϕ n 2 ⁡ ( τ ) ⁢ b n d ⁡ ( k ) ⁢ ( b ~ n ⁡ ( k ) τ - b n d ⁡ ( k ) ) H , wherein ϕ n 2 (τ) is a gain factor, defined as ϕ n 2 (τ)=λϕ n 2 (τ−1)+| b n d ( k )| 2 , λ is a forgetting factor, Û n (k) τ H is an n-th diagonal element of a τ-th iteration of the diagonal matrix, b n d (k) is an n-th element of the plurality of desired physical coefficients, and {tilde over (b)} n (k) τ is an n-th element of a τ-th iteration of the plurality of measured physical coefficients. 9. The sound device of claim 1 , wherein the signal processor is further configured to, when updating the plurality of drive signals, compute a drive signal update σ* such that an energy level of the drive signal update σ* is limited with an upper bound, wherein the energy level of the drive signal update σ* is computed as a square value of the drive signal update σ*. 10. The sound device of claim 9 , wherein the signal processor is further configured to, when updating the plurality of drive signals, compute the drive signal update σ* as: σ

Assignees

Inventors

Classifications

  • the noise being echo, reverberation of the speech · CPC title

  • Processing in the frequency domain · CPC title

  • Filtering, e.g. Kalman filters or special analogue or digital filters · CPC title

  • Rooms, e.g. ANC inside a room, office, concert hall or automobile cabin · CPC title

  • G10K11/178Primary

    by electro-acoustically regenerating the original acoustic waves in anti-phase · CPC title

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What does patent US10199032B2 cover?
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…
Who is the assignee on this patent?
Huawei Tech Co Ltd
What technology area does this patent fall under?
Primary CPC classification G10K11/178. Mapped technology areas include Physics.
When was this patent published?
Publication date Tue Feb 05 2019 00:00:00 GMT+0000 (Coordinated Universal Time) (B2). Legal status and post-grant events are not shown on this page.
What related patents are in patentsdb?
We list 1 related publication on this page (citations in our corpus or others sharing the same primary CPC).