Adaptive beamformer, sidelobe canceller, handsfree speech communication device

1. An adaptive beamformer unit comprising:

a filtered sum beamformer for processing input audio signals (u 1 , u 2 ) from an array of respective microphones, and for forming, as an output, a first audio signal (z) predominantly corresponding to sound from a desired audio source by filtering, with a first adaptive filter (f 1 (-t)), a first one of the input audio signals (u 1 ) and by filtering, with a second adaptive filter (f 2 (-t)), a second one of the input audio signals (u 2 ), the coefficients of the first filter (f 1 (-t)) and the second filter (f 2 (-t)) being adaptable with a first step size (α 1 ) and a second step size (α 2 ), respectively;

noise measure derivation means for deriving, from the input audio signals (u 1 , u 2 ), a first noise measure (x 1 ) and a second noise measure (x 2 ); and

an updating unit for determining the first and the second step size (α 1 , α 2 ), respectively, with an equation comprising, in a denominator, the first noise measure (x 1 ) for the first step size (α 1 ) or the second noise measure (x 2 ) for the second step size (α 2 ), respectively.

2. The adaptive beamformer unit as claimed in claim 1 , wherein the noise measure derivation means derives the first noise measure (x 1 ) from the first input audio signal (u 1 ) by subtracting a desired sound measure (m 1 ) of the sound from the desired audio source as picked up by the first microphone, and derives the second noise measure (x 2 ) from the second input audio signal (u 2 ) by subtracting a second desired sound measure (m 2 ) of the sound from the desired audio source as picked up by the second microphone.

3. The adaptive beamformer unit as claimed in claim 2 , wherein the equation to obtain the first and second step size (α 1 , α 2 ), respectively, equals:

α m [f,t]=βP zz [f,t ]/( P zz [f,t]+γP x m x m [f,t ]),

in which m is an index indicating which of the first or second adaptive filters (f 1 (-t), f 2 (-t), respectively) is adapted with the resulting step size α m , f denotes a frequency, t a time instant, z the first audio signal, x m is the first or the second noise measure, respectively, P ss denotes an equation to obtain a power of the signal identified in its subscript s, and β and γ are predetermined constants.

4. The adaptive beamformer unit as claimed in claim 1 , wherein the first noise measure (x 1 ) and the second noise measure (x 2 ) are determined from respective linear combinations of the input audio signals (u 1 , u 2 ).

5. The adaptive beamformer unit as claimed in claim 1 , wherein said adaptive beamformer unit further comprises a scaling factor determining unit for determining a single scale factor (S) for scaling the step size (α 1 , α 2 , respectively) of both the first filter (f 1 (-t)) and the second filter (f 2 (-t)) of the beamformer, the scale factor (S) being determined on the basis of an amount of speech leakage and/or uncorrelated noise.

6. The adaptive beamformer unit as claimed in claim 1 , wherein said adaptive beamformer unit receives position data from an audio-based speaker tracker for determining a position in space of a speaker based on the speaker's speech and/or a video-based speaker tracker for determining a position in space of a speaker based on a captured image, in which the first filter (f 1 (-t)) and the second filter (f 2 (-t)) coefficients are initially determined on the basis of the position determined by the audio-based speaker tracker and/or video-based speaker tracker.

7. A sidelobe canceller comprising:

a filtered sum beamformer for processing input audio signals (u 1 , u 2 ) from an array of respective microphones, an for forming, as an output, first audio signal (z) predominantly corresponding to sound a desire audio source by filtering, with a first a filter (f 1 (-)), first one of the input audio signals (u 1 ) and by filtering, with a second adaptive filter (f 2 (-t)), a second one of the input audio signals (u 2 ), the coefficients of the first filter (f 1 (-t)) an the second filter (f 2 (-t)) being adaptable with a first step size (αl) and a second step size (α 2 ), respectively;

an adaptive noise estimator for deriving an estimated noise signal (y) by filtering the first and the second noise measures (x 1 , x 2 ) derived from the input audio signals (u 1 , u 2 ) with a second set of adaptable filters (g 1 , g 2 );

a subtracter for subtracting, the estimated noise signal (y) from the first audio signal (z) to obtain a noise-cleaned second audio signal (r); and

an updating unit for determining the first and second step size (α 1 , α 2 ), respectively, with an equation comprising an amplitude measure of the second audio signal (r) and, in a denominator, the first noise measure (x 1 ) for the first step size (α 1 ) or the second noise measure (x 2 ) for the second step size (α 2 ), respectively.

8. The sidelobe canceller as claimed in claim 7 , wherein the equation to obtain a step size equals:

α m =βP rr [f,t ]/( P rr [f,t ]+γ P v m v m [f,t ]),

in which m is an index indicating which of the first or second adaptive filters (f 1 (-t), f 2 (-t)) is adapted with the resulting step size α m , f denotes a frequency, t a time instant, r the second audio signal, v m is a measure of noise picked up by the corresponding m-th microphone, the noise-cleaned second audio signal (r) as measure of the sound from the desired audio source being subtracted from the respective input signal (u 1 , u 2 ) to obtain the noise measure v m , P denotes an equation to obtain the power of a signal, and β and γ are predetermined constants.

9. The sidelobe canceller as claimed in claim 7 , wherein said sidelobe canceller further comprises a scaling factor determining unit for determining a single scale factor (S) for scaling the step size (α 1 , α 2 , respectively) of both the first filter (f 1 (-t)) and the second filter (f 2 (-t)) of the beamformer, the scale factor (S) being determined on the basis of an amount of speech leakage and/or uncorrelated noise.

10. A handsfree speech communication system comprising an adaptive beamformer unit as claimed in claim 1 .

11. A portable speech communication device comprising at least two microphones to yield input audio signals (u 1 , u 2 ), and further comprising an adaptive beamformer unit as claimed in claim 1 to process the input audio signals (u 1 , u 2 ).

12. A voice control unit comprising an adaptive beamformer unit as claimed in claim 1 , and further comprising speech analysis means for recognizing voice commands.

13. A consumer apparatus comprising a voice control unit as claimed in claim 12 .

14. A method of adaptive beamforming, comprising the steps of:

a) filtering a first input audio signal (u 1 ) from a first microphone ( 101 ) with a first adaptive filter (f 1 (-t)), filtering a second input audio signal (u 2 ) from a second microphone ( 103 ) with a second adaptive filter (f 2 (-t)), and summing the filtered input audio signals to yield a first audio signal (z) predominantly corresponding to sound from a desired audio source;

b) deriving a first noise measure (x 1 ) and a second noise measure (x 2 ) from the input audio signals (u 1 , u 2 ); and

c) adapting the coefficients of the first filter (f 1 (-t)) and the second filter (f 2 (-t)) with a first step size (α 1 ) and a second step size (α 2 ), respectively, said step sizes result from an equation comprising, in a denominator, the first noise measure (x 1 ) for the first step size (α 1 ) or the second noise measure (x 2 ) for the second step size (α 2 ), respectively.

15. A non-transitory computer-readable medium having stored therein a computer program comprising code which, when loaded into a processor, causes the processor to perform the method as claimed in claim 14 .