Rendering system

Patent metadata

Field	Value
Publication number	US-10659901-B2
Application number	US-201815920914-A
Country	US
Kind code	B2
Filing date	Mar 14, 2018
Priority date	Sep 25, 2015
Publication date	May 19, 2020
Grant date	May 19, 2020

Abstract

A rendering system including a plurality of loudspeakers, at least one microphone and a signal processing unit. The signal processing unit is configured to determine at least some components of a loudspeaker-enclosure-microphone transfer function matrix estimate describing acoustic paths between the plurality of loudspeakers and the at least one microphone using a rendering filters transfer function matrix using which a number of virtual sources is reproduced with the plurality of loudspeakers.

First claim

The invention claimed is:

1. A rendering system, comprising:
plurality of loudspeakers;
at least one microphone;
a signal processing unit;
wherein using a rendering filters transfer function matrix a number of virtual sources is reproduced with the plurality of loudspeakers; and
wherein the signal processing unit is configured to determine at least some components of a loudspeaker-enclosure-microphone transfer function matrix estimate describing acoustic paths between the plurality of loudspeakers and the at least one microphone using said rendering filters transfer function matrix;
wherein the signal processing unit is configured to estimate at least some components of a source-specific transfer function matrix describing acoustic paths between the number of virtual sources and the at least one microphone; and

wherein the processing unit is configured to determine the loudspeaker-enclosure-microphone transfer function matrix estimate using the estimated source-specific signal transfer function matrix; wherein the signal processing unit is configured to determine at least some components of the loudspeaker-enclosure-microphone transfer function matrix estimate based on the equation
Ĥ=Ĥ S H D + ,

wherein Ĥ represents the loudspeaker-enclosure-microphone transfer function matrix estimate, wherein Ĥ S represents the estimated source-specific transfer function matrix, wherein H D represents the rendering filters transfer function matrix, and wherein H D + represents an approximate inverse of the rendering filters' transfer function matrix H D .

2. The rendering system according to claim 1 , wherein the signal processing unit is configured to adaptively estimate the source-specific transfer function matrix by minimizing a cost function derived from a difference between a recorded signal of the at least one microphone and an estimated signal of the at least one microphone obtained using the estimated source-specific transfer function matrix.

3. The rendering system according to claim 1 , wherein the signal processing unit is configured to determine the components of the loudspeaker-enclosure-microphone transfer function matrix estimate which are sensitive to a column space of the rendering filters transfer function matrix.

4. The rendering system according to claim 1 , wherein in response to a change of at least one out of a number of virtual sources and a position of at least one of the virtual sources, the signal processing unit is configured to update at least some components of the loudspeaker-enclosure-microphone transfer function matrix estimate using a rendering filters transfer function matrix corresponding to the changed virtual sources.

5. The rendering system according to claim 1 , wherein the signal processing unit is configured to update at least some components of the loudspeaker-enclosure-microphone transfer function matrix estimate based on the equation
Ĥ (κ|κ)= Ĥ ⊥ (κ|κ−1)+ Ĥ S (κ|κ) H D + (κ)

wherein κ−1 denotes a previous time interval, wherein κ denotes a current time interval, wherein between the previous time interval and the current time interval at least one out of a number of virtual sources and a position of at least one of the virtual sources is changed, wherein Ĥ(κ|κ) represents a loudspeaker-enclosure-microphone transfer function matrix estimate, Ĥ ⊥ (κ|κ−1) represents components of the loudspeaker-enclosure-microphone transfer function matrix estimate which are not sensitive to the column space of the rendering filters transfer function matrix, Ĥ S (κ|κ) represents an estimated source-specific transfer function matrix, and wherein H D + (κ) represents an inverse rendering filters transfer function matrix.

6. The rendering system according to claim 4 , wherein the signal processing unit is configured to update at least some components of the loudspeaker-enclosure-microphone transfer function matrix estimate based on the equation
Ĥ (κ|κ)= Ĥ (κ|κ−1)+( Ĥ S (κ|κ)− Ĥ S (κ|κ−1)) H D + (κ)

in order to reduce an average load of the signal processing unit;
wherein κ−1 denotes a previous time interval, wherein κ denotes a current time interval, wherein between the current time interval and the previous time interval at least one out of a number of virtual sources and a position of at least one of the virtual sources is changed, wherein Ĥ(κ|κ) represents a loudspeaker-enclosure-microphone transfer function matrix estimate, wherein Ĥ(κ|κ−1) represents a loudspeaker-enclosure-microphone transfer function matrix estimate, Ĥ S (κ|κ) represents an estimated source-specific transfer function matrix, wherein Ĥ(κ|κ−1) represents a loudspeaker-enclosure-microphone transfer function matrix estimate, and wherein H D + (κ) represents an inverse rendering filters transfer function matrix.

7. The rendering system according to claim 4 , wherein the signal processing unit is configured to update at least some components of the loudspeaker-enclosure-microphone transfer function matrix estimate based on the distributedly evaluated equation
Ĥ (κ|κ)= Ĥ (κ−1|κ−2)+ Ĥ S Δ (κ−1) H D + (κ−1)

as part of an initialization of a following interval's estimated source-specific transfer function matrix by
Ĥ S (κ+1|κ)=( Ĥ (κ−1|κ−2)+ Ĥ S Δ (κ−1) H D + (κ−1)) H D (κ+1)+ Ĥ S Δ (κ) H T (κ,κ+1)

in order to reduce a peak load of the signal processing unit;
wherein κ−2 denotes a second previous time interval, wherein κ−1 denotes a previous time interval, wherein κ denotes a current time interval, wherein κ−1 denotes a following time interval, wherein between the time intervals at least one out of a number of virtual sources and a position of at least one of the virtual sources is changed, wherein Ĥ(κ|κ−1) represents a loudspeaker-enclosure-microphone transfer function matrix estimate, Ĥ S (κ+1|κ) represents an estimated source-specific transfer function matrix, wherein Ĥ(κ−1|κ−2) represents a loudspeaker-enclosure-microphone transfer function matrix estimate, wherein Ĥ S Δ (κ−1) represents an update of an estimated source-specific transfer function matrix, H D + (κ−1) represents an inverse rendering filters transfer function matrix, H D (κ+1) represents a rendering filters transfer function matrix, Ĥ S Δ (κ) represents an update of an estimated source-specific transfer function matrix, and wherein H T (κ,κ+1) represents a transition transform matrix which describes an update of an estimated source-specific transfer function matrix of the current time interval to the following time interval, such that only a contribution of Ĥ S Δ (κ)H T (κ,κ+1) is computed between two time intervals.

8. The rendering system according to claim 1 , wherein a number of virtual sources is smaller than a number of loudspeakers.

9. The rendering system according to claim 1 , wherein the signals of the virtual sources are statistically independent.

10. A method, comprising:
determining at least some components of a loudspeaker-enclosure-microphone transfer function matrix estimate describing acoustic paths between a plurality of loudspeakers and at least one microphone using a rendering filters transfer function matrix, wherein using said rendering filters transfer function matrix a number of virtual sources is reproduced with the plurality of loudspeakers; and

estimating at least some components of a source-specific transfer function matrix describing acoustic paths between the number of virtual sources and the at least one microphone, wherein the loudspeaker-enclosure-microphone transfer function matrix estimate is determined using the estimated source-specific signal transfer function matrix;
wherein at least some componen

Assignees

• Fraunhofer Ges Forschung

Inventors

• Hofmann Christian
• Kellermann Walter

Classifications

• H04S2420/13

  Application of wave-field synthesis in stereophonic audio systems · CPC title

• H04S2420/01

  Enhancing the perception of the sound image or of the spatial distribution using head related transfer functions [HRTF's] or equivalents thereof, e.g. interaural time difference [ITD] or interaural level difference [ILD] · CPC title

• H04S2400/11

  Positioning of individual sound objects, e.g. moving airplane, within a sound field (H04S2420/13 takes precedence) · CPC title

• H04S2420/11

  Application of ambisonics in stereophonic audio systems · CPC title

• H04R5/02

  Spatial or constructional arrangements of loudspeakers · CPC title