Apparatus and method for listening room equalization using a scalable filtering structure in the wave domain

Patent metadata

Field	Value
Publication number	US-9338576-B2
Application number	US-201414226296-A
Country	US
Kind code	B2
Filing date	Mar 26, 2014
Priority date	Sep 27, 2011
Publication date	May 10, 2016
Grant date	May 10, 2016

Abstract

An apparatus for listening room equalization is provided. A system identification adaptation unit is configured to adapt a first loudspeaker-enclosure-microphone system identification to obtain a second loudspeaker-enclosure-microphone system identification. A filter adaptation unit is configured to adapt a filter based on the second loudspeaker-enclosure-microphone system identification a predetermined loudspeaker-enclosure-microphone system identification. A filter includes a plurality of subfilters each of which receive one or more of the transformed loudspeaker signals. Each of the subfilters is adapted to generate one of a plurality of filtered loudspeaker signals based on the one or more received loudspeaker signals. At least one of the subfilters is arranged to couple the at least two received loudspeaker signals to generate one of the plurality of the filtered loudspeaker signals. At least one of the subfilters has a number of the received loudspeaker signals that is smaller than a total number of the plurality of transformed loudspeaker signals.

First claim

The invention claimed is:

1. An apparatus for listening room equalization, wherein the apparatus is adapted to receive a plurality of loudspeaker input signals, and wherein the apparatus comprises:
a first transform unit for transforming the at least two loudspeaker input signals from a time domain to a wave domain to acquire a plurality of transformed loudspeaker signals,
a system identification adaptation unit for adapting a first loudspeaker-enclosure-microphone system identification to acquire a second loudspeaker-enclosure-microphone system identification, wherein the first and the second loudspeaker-enclosure-microphone system identification identify a loudspeaker-enclosure-microphone system comprising a plurality of loudspeakers and a plurality of microphones,
a filter, wherein the filter comprises a plurality of subfilters for generating a plurality of filtered loudspeaker signals,
an inverse transform unit for transforming the plurality of filtered loudspeaker signals from the wave domain to the time domain to acquire filtered time-domain loudspeaker signals and for feeding the filtered time-domain loudspeaker signals into the plurality of loudspeakers of the loudspeaker-enclosure-microphone system,
a filter adaptation unit for adapting the filter based on the second loudspeaker-enclosure-microphone system identification and based on a predetermined loudspeaker-enclosure-microphone system identification, wherein the system identification adaptation unit is configured to adapt the first loudspeaker-enclosure-microphone system identification based on an error indicating a difference between a plurality of transformed microphone signals and a plurality of estimated microphone signals, wherein the plurality of transformed microphone signals and the plurality of estimated microphone signals depend on the plurality of the filtered loudspeaker signals, wherein the filter is defined by a first matrix {tilde over (G)}(n), wherein the first matrix {tilde over (G)}(n) comprises a plurality of first matrix coefficients, wherein the filter adaptation unit is configured to adapt the filter by adapting the first matrix {tilde over (G)}(n), and wherein the filter adaptation unit is configured to adapt the first matrix {tilde over (G)}(n) by setting one or more of the plurality of first matrix coefficients to zero,
a second transform unit for receiving a plurality of microphone signals as received by the plurality of microphones and for transforming a plurality of microphone signals of the loudspeaker-enclosure-microphone system from a time domain to a wave domain to acquire the plurality of transformed microphone signals, and
a loudspeaker-enclosure-microphone system estimator for generating the plurality of estimated microphone signals based on the first loudspeaker-enclosure-microphone system identification and based on the plurality of the filtered loudspeaker signals,
wherein each subfilter of the subfilters is arranged to receive one or more of the transformed loudspeaker signals as received loudspeaker signals of said subfilter, and wherein each subfilter of the subfilters is furthermore adapted to generate one of the plurality of filtered loudspeaker signals based on the one or more received loudspeaker signals of said subfilter,
wherein at least one subfilter of the subfilters is arranged to receive at least two of the transformed loudspeaker signals as the received loudspeaker signals of said subfilter, and is furthermore arranged to couple the at least two received loudspeaker signals of said subfilter to generate one of the plurality of the filtered loudspeaker signals of said subfilter,
wherein at least one subfilter of the subfilters comprises a number of the received loudspeaker signals of said subfilter that is smaller than a total number of the plurality of transformed loudspeaker signals, the number of the received loudspeaker signals of said subfilter being one or greater than one, and wherein, when the number of the received loudspeaker signals of a subfilter of the at least one of the subfilters is greater than one, only the received loudspeaker signals of the subfilter of the at least one of the subfilters are coupled to generate the one of the plurality of the filtered loudspeaker signals.

2. An apparatus according to claim 1 ,
wherein the filter adaptation unit is configured to determine a filter coefficient for each pair of at least three pairs of a signal pair group to acquire a filter coefficients group, the signal pair group comprising all loudspeaker signal pairs of one of the transformed loudspeaker signals and one of the filtered loudspeaker signals, wherein the filter coefficients group comprises fewer filter coefficients than the signal pair group comprises loudspeaker signal pairs, and
wherein the filter adaptation unit is configured to adapt the filter by replacing filter coefficients of the filter by at least one of the filter coefficients of the filter coefficients group.

3. An apparatus according to claim 1 ,
wherein the filter adaptation unit is configured to determine a filter coefficient for each pair of a signal pair group to acquire a first filter coefficients group, the signal pair group comprising all loudspeaker signal pairs of one of the transformed loudspeaker signals and one of the filtered loudspeaker signals,
wherein the filter adaptation unit is configured to select a plurality of filter coefficients from the first filter coefficients group to acquire a second filter coefficients group, the second filter coefficients group comprising fewer filter coefficients than the first filter coefficients group, and
wherein the filter adaptation unit is configured to adapt the filter by replacing filter coefficients of the filter by at least one of the filter coefficients of the second filter coefficients group.

4. An apparatus according to claim 1 , wherein all subfilters of the filter receive the same number of transformed loudspeaker signals.

5. An apparatus according to claim 1 , wherein the filter adaptation unit is configured to adapt the filter based on the equation
{tilde over (H)} ( n ) {tilde over (G)} ( n )= {tilde over (H)} (0)
wherein {tilde over (H)}(n) is a second matrix indicating the second loudspeaker-enclosure-microphone system identification, and
wherein {tilde over (H)} (0) is a third matrix indicating the predetermined loudspeaker-enclosure-microphone system identification.

6. An apparatus according to claim 5 , wherein the second matrix {tilde over (H)}(n) comprises a plurality of second matrix coefficients, and wherein the system identification adaptation unit is configured to determine the second matrix {tilde over (H)}(n) by setting one or more of the plurality of second matrix coefficients to zero.

7. An apparatus according to claim 1 ,
wherein the apparatus furthermore comprises an error determiner for determining the error {tilde over (e)}(n) indicating the difference between the plurality of transformed microphone signals and the plurality of estimated microphone signals by applying the formula
{tilde over (e)} ( n )= {tilde over (d)} ( n )− {tilde over (y)} ( n )

to determine the error, and
wherein the error determiner is arranged to feed the determined error into the system identification adaptation unit.

8. A method for listening room equalization comprising:
receiving a plurality of loudspeaker input signals,
transforming the at least two loudspeaker input signals from a time domain to a wave domain to acquire a plurality of transformed loudspeaker signals,
adapting a first loudspeaker-enclosure-microphone system identification to acquire a second loudspeaker-enclosure-microphone system identification, wherein the first and the second loudspeaker-enclosure-m

Assignees

• Fraunhofer Ges Forschung

Inventors

• Schneider Martin
• Kellermann Walter

Classifications

• H04S2420/13

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

• H04S7/301 Primary

  Automatic calibration of stereophonic sound system, e.g. with test microphone · CPC title

• H04S2400/09

  Electronic reduction of distortion of stereophonic sound systems · CPC title

• H04S2420/11

  Application of ambisonics in stereophonic audio systems · CPC title