Method and device for separating acoustic signals
View Patent ↗In a method of separating acoustic signals from a plurality of sound sources comprising the following steps: disposing two microphones (MIK 1 , MIK 2 ) at a predefined distance (d) from one another; picking up the acoustic signals with both microphones (MIK 1 , MIK 2 ) and generating associated microphone signals (m 1 , m 2 ); and separating the acoustic signal of one of the sound sources (SI) from the acoustic signals of the other sound sources (S 2 ) on the basis of the microphone output signals (m 1 , m 2 ), the proposed separation step comprises the following steps: applying a Fourier transform to the microphone output signals in order to determine their frequency spectra (M 1 , M 2 ); determining the phase difference between the two microphone output signals (m 1 , m 2 ) for every frequency component of their frequency spectra (M 1 , M 2 ); determining the angle of incidence of every acoustic signal allocated to a frequency of the frequency spectra (M 1 , M 2 ) on the basis of the relative phase angle and the frequency; generating a signal spectrum ( 5 ) of a signal to be output by correlating one of the two frequency spectra (M 1 , M 2 ) with a filter function which is selected so that acoustic signals from an area around a preferred angle of incidence are amplified relative to acoustic signals from outside this area; and applying an inverse Fourier transform to the resultant signal spectrum.
1. Method of separating acoustic signals from a plurality of sound sources (S 1 , S 2 ), comprising the following steps:
disposing two microphones (MIK 1 , MIK 2 ) at a predefined distance (d) from one another;
picking up the acoustic signals with both microphones (MIK 1 , MIK 2 ) and generating associated microphone signals (m 1 , m 2 ); and
separating the acoustic signal of one of the sound sources (S 1 ) from the acoustic signals of the other sound sources (S 2 ) on the basis of the microphone signals (m 1 , m 2 ),
in which the separation step comprises the following steps:
applying a Fourier transform to the microphone signals in order to determine their frequency spectra (M 1 , M 2 );
determining the phase difference (φ) between the two microphone signals (m 1 , m 2 ) for every frequency component of their frequency spectra (M 1 , M 2 );
determining the angle of incidence (θ) of every acoustic signal allocated to a frequency of the frequency spectra (M 1 , M 2 ) on the basis of the phase difference (φ) and the frequency;
generating a signal spectrum (S) of a signal to be output by correlating one of the two frequency spectra (M 1 , M 2 ) with a filter function (F θ 0 ) which is selected so that acoustic signals from an area (γ 3db ) around a preferred angle of incidence (θ 0 ) are amplified relative to acoustic signals from outside this area (γ 3db ); and
applying an inverse Fourier transform to the resultant signal spectrum, characterised in that the filter function (F θ 0 ) is dependent on the angle of incidence θ and has a maximum at the preferred angle of incidence (θ 0 ) when the angle of incidence θ is varied, and the correlation of the filter function (F θ 0 ) with one of the two frequency spectra comprises multiplying the same.
2. Method as claimed in claim 1 , characterised in that the filter function (F θ 0 ) is expressed as follows:
F θ 0 ( f,T )= Z (θ−θ 0 )+ DΔ 2 f Z (θ−θ 0 )
in which
f is the respective frequency
T is the instant at which the frequency spectra (M 1 , M 2 ) are determined
Z(θ−θ 0 ) is an allocation function with a maximum at θ 0
D≧0 is a diffusion constant and
Δ 2 is a discrete diffusion operator.
3. Method as claimed in claim 2 , characterised in that the allocation function (Z) is expressed as follows:
Z
(
ϑ
-
ϑ
0
)
=
(
1
+
cos
(
ϑ
-
ϑ
0
)
2
)
n
where
n
>
0.
4. Method as claimed in claim 1 , characterised in that the angle of incidence θ is determined by the equation
θarc cos(x(f,T))
with
x ( f,T )φ, c/ 2π fd
where
φ is the phase difference between the two microphone signal components (m 1 , m 2 )
c is the acoustic velocity
f is the frequency of the acoustic signal component and
d is the predefined distance of the two microphones (MIK 1 , MIK 2 ).
5. Method as claimed in claim 4 , characterised in that it additionally incorporates the following step:
limiting the value of x(f,T) to the interval [−1,1].
6. Method as claimed in claim 5 , characterised in that it additionally incorporates the following step:
reducing signal components whose value of x(f,T) lay outside of the interval [−1,1] prior to limitation.
7. Device for implementing the method as claimed in claim 1 , comprising:
two microphones (MIK 1 , MIK 2 );
a sampling and Fourier transform unit ( 20 ) connected to the microphones for discretizing and digitising the microphone signals (m 1 , m 2 ) and applying a Fourier transform to them;
a calculating unit ( 30 ) connected to the sampling and Fourier transform unit ( 20 ) for calculating the angle of incidence (θ) of every acoustic signal component; and
at least one signal generator ( 40 ) connected to the calculating unit ( 30 ) for outputting the separated acoustic signal, at least one signal generator ( 40 ) having means for multiplying one of the Fourier transformed frequency spectra (M 1 , M 2 ) by a filter function (F θ 0 ) which is dependent on θ and has a maximum at a preferred angle of incidence (θ 0 ) when θ is varied.
8. Device as claimed in claim 7 , characterised in that the distance (d) between the microphones satisfies the equation:
d<c/ 4 f A
where c is the acoustic velocity and f A is the sampling frequency of the stereo sampling and Fourier transform unit ( 20 ).
9. Device as claimed in claim 7 , characterised in that the device has a signal generator ( 40 ) for every sound source (S 1 , S 2 ) to be separated.