Electrical audio signal processing systems and devices – Directive circuits for microphones
Reexamination Certificate
1998-03-16
2001-11-13
Mei, Xu (Department: 2644)
Electrical audio signal processing systems and devices
Directive circuits for microphones
C381S091000, C381S066000, C381S094100, C381S122000
Reexamination Certificate
active
06317501
ABSTRACT:
BACKGROUND THE INVENTION
Field of the Invention
The present invention relates to a microphone array apparatus which has an array of microphones in order to detect the position of a sound source, emphasize a target sound and suppress noise.
The microphone array apparatus has an array of a plurality of omnidirectional microphones and equivalently define a directivity by emphasizing a target sound and suppressing noise. Further, the microphone array apparatus is capable of detecting the position of a sound source on the basis of a relationship among the phases of output signals of the microphones. Hence, the microphone array apparatus can be applied to a video conference system in which a video camera is automatically oriented towards a speaker and a speech signal and a video signal can concurrently be transmitted. In addition, the speech of the speaker can be clarified by suppressing ambient noise. The speech of the speaker can be emphasized by adding the phases of speech components. It is now required that the microphone array apparatus can stably operate.
If the microphone array apparatus is directed to suppressing noise, filters are connected to respective microphones and filter coefficients are adaptively or fixedly set so as to minimize noise components (see, for example, Japanese Laid-Open Patent Application No. 5-111090). If the microphone array apparatus is directed to detecting the position of a sound source, the relationship among the phases of the output signals of the microphones is detected, and the distance to the sound source is detected (see, for example, Japanese Laid-Open Patent Application Nos. 63-177087 and 4-236385).
An echo canceller is known as a device which utilizes the noise suppressing technique. For example, as shown in
FIG. 1
, a transmit/receive interface
202
of a telephone set is connected to a network
203
. An echo canceller is connected between a microphone
204
and a speaker
205
. A speech of a speaker is input to the microphone
204
. A speech of a speaker on the other (remote) side is reproduced through the speaker
205
. Hence, a mutual communication can take place.
A speech transferred from the speaker
205
to the microphone
204
, as indicated by a dotted line shown in
FIG. 1
forms an echo (noise) to the other-side telephone set. Hence, the echo canceller
201
is provided that includes a subtracter
206
, an echo component generator
207
and a coefficient calculator
208
. Generally, the echo generator
207
has a filter structure which produces an echo component from the signal which drives the speaker
205
. The subtracter
206
subtracts the echo component from the signal from the microphone
204
. The coefficient calculator
208
controls the echo generator
207
to update the filter coefficients so that the residual signal from the subtracter
206
is minimized.
The updating of the filter coefficients c1, c2, . . . , cr of the echo component generator
207
having the filter structure can be obtained by a known maximum drop method. For example, the following evaluation function J is defined based on an output signal e (the residual signal in which the echo component has been subtracted) of the subtracter
206
:
J=e
2
(1)
According to the above evaluation function, the filter coefficients c1, c2, . . . , cr are updated as follows:
(
c1
c2
⋮
cr
)
=
=
=
(
c1
old
c2
old
⋮
cr
old
)
+
a
*
(
e
/
f
norm
)
*
(
f
(
1
)
f
(
2
)
⋮
f
(
r
)
)
(
2
)
where 0.0<&agr;<0.5
f
norm
=(f(1)
2
+f(2)
2
+ . . . f(r)
2
)
½
(3)
In the above expressions, a symbol “*” denotes multiplication, and “r” denotes the filter order. Further, f(1), . . . , f(r) respectively denote the values of a memory (delay unit) of the filter (in other words, the output signals of delay units each of which delays the respective input signal by a sample unit). A symbol “f
norm
” is defined as equation (3), and a symbol “&agr;” is a constant, which represents the speed and precision of convergence of the filter coefficients towards the optimal values.
The echo canceller
201
has filter orders as many as
100
. Hence, another echo canceller using a microphone array as shown in
FIG. 2
is known. There are provided an echo canceller
211
, a transmit/receive interface
212
, microphones
214
-
1
-
214
-n forming a microphone array, a speaker
215
, a subtracter
216
, filters
217
-
1
-
217
-n, and a filter coefficient calculator
218
.
In the structure shown in
FIG. 2
, acoustic components from the speaker
215
to the microphones
214
-
1
-
214
-n are propagated along routes indicated by broken lines and serve as echoes. Hence, the speaker
215
is a noise source. The updating control of the filter coefficients c11, c12, . . . , c1r, . . . , cn1, cn2, . . . , cnr in the case where the speaker does not make any speech is expressed by using the evaluation function (1) as follows:
[
c11
c12
⋮
c1r
]
=
[
c11
old
c12
old
⋮
c1r
old
]
-
a
*
(
e
/
f1
norm
)
*
[
f1
(
1
)
f1
(
2
)
⋮
f1
(
r
)
]
(
4
)
[
cp1
cp2
⋮
cpr
]
=
[
cp1
old
cp2
old
⋮
cpr
old
]
+
a
*
(
e
/
fp
norm
)
*
[
fp
(
1
)
fp
(
2
)
⋮
fp
(
r
)
]
(
5
)
where p=2, 3, . . . , n
The equation (4) relates to a case where one of the microphones
214
-
1
-
214
-n, for example, the microphone
214
-
1
is defined as a reference microphone, and indicates the filter coefficients c11, c12, . . . , c1r of the filter
217
-
1
which receives the output signal of the above reference microphone
214
-
1
. The equation (5) relates to the microphones
214
-
2
-
214
-n other than the reference microphones, and indicates the filter coefficients c21, c22, . . . , c2r, . . . , cn1, cn2, . . . , cnr. The subtracter
216
subtracts the output signals
217
-
2
-
217
-n of the microphones
214
-
2
-
214
-n from the output signal
217
-
1
of the reference microphone
214
-
1
.
FIG. 3
is a block diagram for explaining a conventional process of detecting the position of a sound source and emphasizing a target sound. The structure shown in
FIG. 3
includes a target sound emphasizing unit
221
, a sound source detecting unit
222
, delay units
223
and
224
, a number-of-delayed-samples calculator
225
, an adder
226
, a crosscorrelation coefficient calculator
227
, a position detection processing unit
228
and microphones
229
-
1
and
229
-
2
.
The target sound emphasizing unit
221
includes the delay units
223
and
224
of Z
−da
and Z
−db
, the number-of-delayed-samples calculator
225
and the adder
226
. The sound source position detecting unit
222
includes the crosscorrelation coefficient calculator
227
and the position detection processing unit
228
. The number-of-delayed samples calculator
225
is controlled by the following factors. The crosscorrelation coefficient calculator
227
of the sound source position detecting unit
222
obtains a crosscorrelation coefficient r(i) of output signals a(j) and b(j) of the microphones
229
-
1
and
229
-
2
. The position detection processing unit
228
obtains the sound source position by referring to a value of i, imax, at which the maximum of the crosscorrelation coefficient r(i) can be obtained.
The crosscorrelation coefficient r(i) is expressed as follows:
r(i)=&Sgr;
n
j=1
a(j)*b(j+i) (6)
where &Sgr;
n
j=1
denotes a summation of j=1 to j=n, and i has a relationship −m≦i ≦m. The symbol “m” is a value dependent on the distance between the microphones
229
-
1
and
229
-
2
and the sampling frequency, and is written as follows:
m=[(sampling frequency)*(intermichrophone distance)]/(speed of sound) (7)
where n is the number of samples for a convolutional operation.
The number of delayed samples da of the Z
−da
delay unit
223
and the number of delayed samples db of the Z
−db
delay unit
224
can be obtained as follows from the value imax at which th
Fujitsu Limited
Mei Xu
Rosenman & Colin LLP
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