Microphone system and a hearing device comprising a microphone system

• US 10,631,102 B2
• Filed: 06/08/2018
• Issued: 04/21/2020
• Est. Priority Date: 06/09/2017
• Status: Active Grant

• Current Assignee: Oticon A/S (William Demant Fonden)
• Original Assignee: Oticon A/S (William Demant Fonden)
• Inventors: Jensen, Jesper, De Haan, Jan Mark, Pedersen, Michael Syskind
• Primary Examiner(s): Ojo, Oyesola C
• Application Number:
  US16/003,396
• Publication Number:
  US 20180359572A1
• Time in Patent Office: 683 Days
• CPC Class Codes: H04R 1/406 microphones
• CPC Class Codes: H04R 2420/01 Input selection or mixing for amplifiers or loudspeakers for hearing aids H04R25/43
• CPC Class Codes: H04R 25/405 by combining a plurality of transducers
• CPC Class Codes: H04R 25/407 Circuits for combining signals of a plurality of transducers
• CPC Class Codes: H04R 25/453 electronically
• CPC Class Codes: H04R 25/505 using digital signal processing
• CPC Class Codes: H04R 25/552 Binaural
• CPC Class Codes: H04R 25/554 using a wireless connection, e.g. between microphone and amplifier or using Tcoils

Abstract

A microphone system comprises a multitude of microphones; a signal processor connected to said number of microphones, and being configured to estimate a direction- to and/or a position of the target sound source relative to the microphone system based on a maximum likelihood methodology; and a database Θ comprising a dictionary of relative transfer functions representing direction-dependent acoustic transfer functions from said target signal source to each of said microphones relative to a reference microphone among said microphones, wherein individual dictionary elements of said database Θ of relative transfer functions comprises relative transfer functions for a number of different directions and/or positions relative to the microphone system; and wherein the signal processor is configured to determine one or more of the most likely directions to or locations of said target sound source. The invention may e.g. be used for the hearing aids or other portable audio communication devices.

Claims

1. A microphone system adapted to be worn at an ear of a user, the microphone system comprising

a multitude of M of microphones, where M is larger than or equal to two, adapted for picking up sound from the environment and to provide M corresponding electric input signals x_m(n), m=1, . . . , M, n representing time, the environment sound at a given microphone comprising a mixture of a target sound signal s_m(n) propagated via an acoustic propagation channel from a location of a target sound source, and possible additive noise signals v_m(n) as present at the location of the microphone in question;

a signal processor connected to said number of microphones, and being configured to estimate a direction- to and/or a position of the target sound source relative to the microphone system based on

a maximum likelihood methodology, and

a database Θ

comprising a dictionary of vectors d_θ

, termed RTF-vectors, whose elements are relative transfer functions d_m(k) representing direction-dependent acoustic transfer functions from said target signal source to each of said M microphones (m=1, . . . , M) relative to a reference microphone (m=i) among said M microphones, k being a frequency index, wherein

individual dictionary elements of said database Θ

of RTF vectors d_θ

comprises relative transfer functions for a number of different directions (θ

) and/or positions (θ

, φ

, r) relative to the microphone system;

the signal processor is configured to

determine a posterior probability or a log (posterior) probability of some of or all of said individual dictionary elements, and

determine one or more of the most likely directions to or locations of said target sound source by determining the one or more values among said determined posterior probabilities or said log (posterior) probabilities having the largest posterior probability(ies) or log (posterior) probability(ies), respectively; and

said relative transfer functions d_m(k) of the database Θ

represent direction-dependent filtering effects of the head and torso of the user in the form of direction-dependent acoustic transfer functions from said target signal source to each of said M microphones (m=1, . . . , M) relative to a reference microphone (m=i) among said M microphones.

2. A microphone system according to claim 1 wherein the signal processor is configured to determine a likelihood function or a log likelihood function of some or all of the elements in the dictionary Θ

in dependence of a noisy target signal covariance matrix C_x and a noise covariance matrix C_v.

3. A microphone system according to claim 2 wherein said noisy target signal covariance matrix C_x and said noise covariance matrix C_v are estimated and updated based on a voice activity estimate and/or an SNR estimate, e.g. on a frame by frame basis.

4. A microphone system according to claim 2 wherein said noisy target signal covariance matrix C_x and said noise covariance matrix C_v are represented by smoothed estimates.

5. A microphone system according to claim 4 wherein said smoothed estimates of said noisy covariance matrix Ĉ

_x and/or said noise covariance matrix Ĉ

_v are determined by adaptive covariance smoothing.

6. A microphone system according to claim 5 wherein said adaptive covariance smoothing comprises determining normalized fast and variable covariance measures, {tilde over (ρ

)}(m) and an {tilde over (p)}(m), respectively, of said noisy covariance matrix Ĉ

_X and/or said noise covariance matrix Ĉ

_V, applying a fast ({tilde over (α

)}) and a variable smoothing factor (α

), respectively, wherein said variable smoothing factor α

is set to fast ({tilde over (α

)}) when the normalized covariance measure of the variable estimator deviates from the normalized covariance measure of the variable estimator by more than a constant value ϵ

, and otherwise to slow (α

₀), i.e.

7. A microphone system according to claim 1 wherein the number of microphones M is equal to two, and wherein the signal processor is configured to calculate a log likelihood of at least some of said individual dictionary elements of said database Θ

of relative transfer functions d_m(k) for at least one frequency sub-band k, according to the following expression

8. A microphone system according to claim 1 wherein the signal processor is configured to estimate the posterior probability or the log (posterior) probability of said individual dictionary elements do of said database Θ

comprising relative transfer functions d<sub>θ

,m</sub>(k), m=1, . . . , M, independently in each frequency band k.

9. A microphone system according to claim 1 wherein the signal processor is configured to estimate the posterior probability or the log (posterior) probability of said individual dictionary elements d_θ

of said database Θ

comprising relative transfer functions d<sub>θ

,m</sub>(k), m=1, . . . , M, jointly across some of or all frequency bands k.

13. A microphone system according to claim 1 wherein the database Θ

of RTF vectors d_θ

comprises an own voice look vector.

10. A microphone system comprising:

a multitude of M of microphones, where M is larger than or equal to two, adapted for picking up sound from the environment and to provide M corresponding electric input signals x_m(n), m=1, . . . , M, n representing time, the environment sound at a given microphone comprising a mixture of a target sound signal s_m(n) propagated via an acoustic propagation channel from a location of a target sound source, and possible additive noise signals v_m(n) as present at the location of the microphone in question;

a signal processor connected to said number of microphones, and being configured to estimate a direction- to and/or a position of the target sound source relative to the microphone system based on

a maximum likelihood methodology;

a database Θ

comprising a dictionary of vectors d_θ

, termed RTF-vectors, whose elements are relative transfer functions d_m(k) representing direction-dependent acoustic transfer functions from said target signal source to each of said M microphones (m=1, . . . , M) relative to a reference microphone (m=i) among said M microphones, k being a frequency index, wherein

individual dictionary elements of said database Θ

of RTF vectors d_θ

comprises relative transfer functions for a number of different directions (θ

) and/or positions (θ

, φ

, r) relative to the microphone system; and

the signal processor is configured to

determine a posterior probability or a log (posterior) probability of some of or all of said individual dictionary elements, and

determine one or more of the most likely directions to or locations of said target sound source by determining the one or more values among said determined posterior probabilities or said log (posterior) probabilities having the largest posterior probability(ies) or log (posterior) probability(ies), respectively; and

the signal processor is configured to utilize information not derived from said electric input signals to determine one or more of the most likely directions to or locations of said target sound source.

11. A microphone system according to claim 10 wherein said information comprises information about eye gaze, and/or information about head position and/or head movement.

12. A microphone system according to claim 10 wherein said information comprises information stored in the microphone system, or received, e.g. wirelessly received, from another device, e.g. from a sensor, or a microphone, or a cellular telephone, and/or from a user interface.

14. A hearing device adapted for being won at or in an ear of a user, or for being fully or partially implanted in the head at an ear of the user, the hearing device comprising;

a microphone system comprising

a multitude of M of microphones, where M is larger than or equal to two, adapted for picking up sound from the environment and to provide M corresponding electric input signals x_m(n), m=1, . . . , M, n representing time, the environment sound at a given microphone comprising a mixture of a target sound signal s_m(n) propagated via an acoustic propagation channel from a location of a target sound source, and possible additive noise signals v_m(n) as present at the location of the microphone in question;

a signal processor connected to said number of microphones, and being configured to estimate a direction- to and/or a position of the target sound source relative to the microphone system based on

a maximum likelihood methodology, and

a database Θ

comprising a dictionary of vectors d_θ

, termed RTF-vectors, whose elements are relative transfer functions d_m(k) representing direction-dependent acoustic transfer functions from said target signal source to each of said M microphones (m=1, . . . , M) relative to a reference microphone (m=i) among said M microphones, k being a frequency index, wherein

individual dictionary elements of said database Θ

of RTF vectors d_θ

comprises relative transfer functions for a number of different directions (θ

) and/or positions (θ

, φ

, r) relative to the microphone system; and

the signal processor is configured to

determine a posterior probability or a log (posterior) probability of some of or all of said individual dictionary elements, and

determine one or more of the most likely directions to or locations of said target sound source by determining the one or more values among said determined posterior probabilities or said log (posterior) probabilities having the largest posterior probability(ies) or log (posterior) probability(ies), respectively; and

a beamformer filtering unit operationally connected to at least some of said multitude of microphones and configured to receive said electric input signals, and configured to provide a beamformed signal in dependence of said one or more of the most likely directions to or locations of said target sound source estimated by said signal processor.

15. A hearing device according to claim 14 wherein said signal processor is configured to smooth said one or more of the most likely directions to or locations of said target sound source before it is used to control the beamformer filtering unit.

16. A hearing device according to claim 15 wherein said signal processor is configured to perform said smoothing over one or more of time, frequency and angular direction.

17. A hearing device according to claim 14 comprising a feedback detector adapted to provide an estimate of a level of feedback in different frequency bands, and wherein said signal processor is configured to weight said posterior probability or log (posterior) probability for frequency bands in dependence of said level of feedback.

18. A hearing device according to claim 14 comprising a hearing aid, a headset, an earphone, an ear protection device or a combination thereof.

19. A method of operating a microphone system comprising a multitude of M of microphones, where M is larger than or equal to two, adapted for picking up sound from the environment, the method comprising:

providing M electric input signals x_m(n), m=1, . . . , M, n representing time, each electric input signal representing the environment sound at a given microphone and comprising a mixture of a target sound signal s_m(n) propagated via an acoustic propagation channel from a location of a target sound source, and possible additive noise signals v_m(n) as present at the location of the microphone in question;

estimating a direction- to and/or a position of the target sound source relative to the microphone system based on

said electric input signals;

a maximum likelihood methodology; and

a database Θ

comprising a dictionary of relative transfer functions d_m(k) representing direction-dependent acoustic transfer functions from each of said M microphones (m=1, . . . , M) to a reference microphone (m=i) among said M microphones, k being a frequency index, wherein

the method further comprises

providing that individual dictionary elements of said database Θ

of relative transfer functions d_m(k) comprises relative transfer functions for a number of different directions (θ

) and/or positions (θ

, φ

, r) relative to the microphone system, where θ

, φ

, and r are spherical coordinates; and

determining a posterior probability or a log (posterior) probability of some of or all of said individual dictionary elements,

determining one or more of the most likely directions to or locations of said target sound source by determining the one or more values among said determined posterior probability or said log (posterior) probability having the largest posterior probability(ies) or log (posterior) probability(ies), respectively, and

reducing computational complexity in determining one or more of the most likely directions to or locations of said target sound source by one or more of dynamically

down sampling,

selecting a subset of the number of dictionary elements,

selecting a subset of the number of frequency channels, and

removing terms in the likelihood function with low importance.

20. A method according to claim 19 wherein the determination of a posterior probability or a log (posterior) probability of some of or all of said individual dictionary elements is performed in two steps,

a first step wherein the posterior probability or the log (posterior) probability is evaluated for a first subset of dictionary elements with a first angular resolution in order to obtain a first rough estimation of the most likely directions, and

a second step wherein the posterior probability or the log (posterior) probability is evaluated for a second subset of dictionary elements around said first rough estimation of the most likely directions so that dictionary elements around the first rough estimation of the most likely directions are evaluated with second angular resolution, wherein the second angular resolution is larger than the first.

21. A method according to claim 19 comprising a smoothing scheme based on adaptive covariance smoothing.

22. A method according to claim 21 comprising adaptive smoothing of a covariance matrix (C_x, C_v) for said electric input signals comprising adaptively changing time constants (τ

_att, τ

_rel) for said smoothing in dependence of changes (Δ

C) over time in covariance of said first and second electric input signals;

wherein said time constants have first values (τ

_att1, τ

_rel1) for changes in covariance below a first threshold value (Δ

C_th1) and second values (τ

_att2, τ

_rel2) for changes in covariance above a second threshold value (Δ

C_th2), wherein the first values are larger than corresponding second values of said time constants, while said first threshold value (Δ

C_th1) is smaller than or equal to said second threshold value (Δ

C_th2).

23. A computer program comprising instructions which, when the program is executed by a computer, cause the computer to carry out the method of according to claim 19.