JPH09140000

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DESCRIPTION JPH09140000
[0001]
BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates mainly
to conference hearing aids used by people with hearing loss.
[0002]
[Prior Art] Among hearing aids who are not so inconvenient in one-on-one conversation if they
wear hearing aids, as in the case of conferences or lectures, when the distance to the speaker
gets far, the other party says There are many people who complain that they can not understand
what they are. There are two major reasons for this.
[0003]
One is the reduction of the input sound pressure level. Generally, the further the distance
between the sound source and the listening position, the lower the input sound pressure level.
The degree to which it falls depends on the directivity of the sound source and the state of the
room's echo, but for example, it is a point sound source (360 in the free sound field (virtual space
with no boundaries reflecting sound, such as walls, floors and ceilings) Assuming that the virtual
sound source emits sound equally in any direction), the sound pressure level attenuates by 6 dB
when the distance between the sound source and the listening point is doubled. In the actual
room, not only the direct sound but also the reflected sound at the boundary is added, though it
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does not attenuate so much, it attenuates about 10 dB from the sound pressure level at 1 m from
5 m away. In the case of a hearing person, an attenuation of about 10 dB in a relatively quiet
room can be sufficiently heard if the speaker speaks normally, but it is a sense of hearing that the
person with hearing loss, especially the inner ear and hearing nerves are impaired. In the case of
sexually deaf people, it becomes quite difficult to hear.
[0004]
Another reason for falling input sound pressure levels is the background noise and room
reverberation. Since background noise, which is background noise generated by air conditioning
in a room, is almost constant, the SN ratio naturally deteriorates as the input sound pressure
level of the speaker's voice decreases. Further, as described above, when the speaker is far, the
direct sound becomes small, and the reverberation component of the room which is the indirect
sound becomes relatively large. Reverberation components as well as background noise interfere
with speech intelligibility. Furthermore, it has been confirmed that in the case of a deaf person,
this background noise and reverberation sound are received to a greater extent than a hearing
person.
[0005]
Conventionally, for the reduction of the input sound pressure level of the former, the deaf person
has only to adjust the gain with the volume controller of the hearing aid. However, adjusting the
gain to the degree that the input sound pressure level changes (for example, the speaker
changes) is very annoying, and even those who say that the volume controller does not move
except when loud and intolerable (for example in a subway car) There are many. As a result, the
voice of a person with a small voice and a person speaking in the distance is difficult to hear. For
example, a device called a compressor can compress the dynamic range of the input signal to
increase the gain for relatively small level inputs. However, a large compression ratio is required
to compensate for the sound pressure drop due to the distance attenuation in this device, and as
a result, the target voice is distorted and the clarity is lost.
[0006]
For the latter effects of noise and reverberation, 1. A method to sharpen the directivity in the
direction of the target sound with a multi-microphone (effective for both noise and reverberation)
2, noise is mainly low-pass component From the low-pass filter with a high-pass filter (effective
for noise only) 3, monitoring the sound input temporally, a method for amplifying only the voice
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(effective for noise only), various noise · reverberation The suppression method is considered.
[0007]
However, these two techniques which are indispensable in the conference hearing of a person
who is deaf, ie, a device which can simultaneously perform automatic compensation of gain and
noise suppression have not been made.
[0008]
An object of the present invention is to provide a hearing aid used by a deaf person at a meeting
etc., which can suppress background noise and can only listen to a target voice at an appropriate
volume and without distortion.
[0009]
SUMMARY OF THE INVENTION According to the present invention, there is provided a hearing
aid for conference according to the present invention, comprising: sound source direction
detecting means for detecting a sound source direction to collect sound; and sound collecting
direction in the direction detected by the sound source direction detecting means. Sound
collecting means and automatic gain adjusting means for automatically adjusting the gain so that
the output level becomes constant each time the directivity of the sound is switched.
[0010]
The sound collection unit is a microphone array, and the sound source direction detection unit
detects a sound source direction by using an arrival time difference of sound waves entering the
microphones at both ends of the microphone array.
[0011]
Furthermore, a delay-and-sum microphone array is used as the microphone array.
[0012]
In the present invention, the direction of the target sound (speaker) is detected, and the sound in
that direction is collected and output.
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When the direction of the target sound changes as the speaker changes, the gain is automatically
adjusted accordingly to keep the output level constant.
[0013]
Also, using the microphone array, the sound source direction is searched for using the time
difference between the arrival of sound waves entering the microphones at both ends, and the
directivity of the microphone is directed to that direction.
The power of the input sound pressure is measured when the directivity is switched, the gain is
determined based on the result, and the gain is automatically amplified or attenuated up to the
gain at an appropriate rise and fall.
[0014]
Furthermore, by using a delay-and-sum microphone array as the microphone array, the delay
amounts of the microphones are made different and the outputs are summed to obtain sharp
directivity.
[0015]
DESCRIPTION OF THE PREFERRED EMBODIMENT FIG. 1 is a block diagram showing the basic
configuration of an embodiment of the present invention.
In this figure, 10 is a sound collecting means, which uses a microphone whose directivity can be
changed.
A sound source direction detecting means 20 detects the direction of the input sound.
An automatic gain adjustment (AGC) unit 30 is triggered by the output of the sound source
direction detection unit 20 at the preceding stage.
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An amplification means 40 amplifies the output of the sound collection means 10 and outputs it
as an appropriate level.
In addition, the output level can be varied. Adjustment of the level once decided is automatically
performed by the AGC means 30.
[0016]
Next, the operation will be described. When the sound production of the speaker is input to the
sound collection means 10, the input direction (sound source direction) is detected by the sound
source direction detection means 20. The directivity of the sound collection means 10 is adjusted
in the detected sound source direction. Sound is collected in this manner, and the output is
amplified by the amplification means 40. When the direction of the sound source changes as the
speaker changes, the output level also fluctuates. Therefore, the AGC means 30 automatically
adjusts the gain of the amplification means 40 so as to keep the output level constant.
[0017]
FIG. 2 is a block diagram showing the configuration of a more specific embodiment of the
present invention. In this embodiment, the sound collecting means 10 comprises a microphone
array 12 consisting of six microphones 11, an amplifier 13 for amplifying the output of each
microphone 11, and a delay and sum circuit 14, respectively. Is further composed of a variable
delay element 15 and an adder 16.
[0018]
The amplification means 40 is composed of an amplifier 41 and a gain multiplier 42. The sound
source direction detecting means 20 uses the time difference between the outputs of the
microphones 12 at both ends of the microphone array 12 (the outputs of the amplifiers 13 at
both ends). The AGC unit 30 is triggered by the output of the sound source direction detection
unit 20, monitors the output level of the amplifier 41, and automatically adjusts the
multiplication factor of the gain multiplier 42 so that the output of the amplification unit 40
becomes constant. 50 shows a headphone.
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[0019]
Next, the operation will be described. First, in the sound source direction detection means 20, the
sound source direction is estimated using the arrival time difference of the sound waves entering
the microphones 11 at both ends of the microphone array 12. For calculating the time difference,
there is a method of calculating the cross correlation function of two input signals and detecting
the peak position. After estimating the sound source direction, direct the directivity in that
direction. As an example, a delay-and-sum microphone array is used so that the directivity is
sharply and electrically variable in the target sound direction.
[0020]
The principle by which a sharp directivity is obtained with the delay-and-sum microphone array
will be described with reference to FIG. 3 (see Sound System and Digital Processing ,
published on March 25, 1995 (first edition) by The Institute of Electronics, Information and
Communication Engineers). It is assumed that the target sound source is at a sufficiently far
distance from the microphone 11 (in principle, infinite distance), and the microphone 11 is
approximated as a plane wave is input. A delay represented by the following equation is added to
each microphone input signal (.theta.L is the target sound direction, c is the speed of sound). Di =
(1-i) τ L where i = 1, 2, 3,..., N (1) τ L = (d sin θ L) / c (2) Assuming that the target sound is
incident from the θ L direction The signals xi (t) received by the microphones 11 are expressed
as follows. xi (t) = x1 (t- (i-1) τL) (3) When a delay of the equation (1) is added to this signal, xi
(t-Di) = x1 (t) (4), All microphone input signals become in-phase signals. That is, all signals from
the sound source direction are in-phase and emphasized. On the other hand, signals from the
other direction remain temporally shifted, and even if N are added, the emphasizing effect is
small, and as a result, sharp directivity in the direction of the target sound can be obtained.
[0021]
At the same time as switching the directivity, the AGC unit 30 is operated by the trigger output
from the sound source direction detection unit 20, the necessary gain is calculated, and
appropriate gain or fall is added and the gain multiplier 42 multiplies the input signal. Do.
[0022]
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FIG. 4 is an example of using the present invention in an actual conference.
In FIG. 4, 100 is a table, and 201 to 210 are attendees of the conference. Now, the attendee 203
is the speaker, and 207 is the wearer of the conference hearing aid of the present invention. The
delay-and-sum microphone array 12 capable of adaptively changing the directivity always has
the directivity in the direction of the speaker 203, and the directivity direction is the sound
source direction detecting means 20 every time the speaker changes. Switched by 2). At the same
time, the necessary gain is calculated in the AGC means 30, and appropriate gain compensation
is performed for the voice of a speaker whose voice is small or distant. The signal is input to, for
example, a magnetic coil type headphone 50 that generates magnetism, and a T mode hearing
aid (a coil for picking up dielectric magnetism is incorporated in addition to the microphone, and
switching is used, respectively). It is called mode. Listen at).
[0023]
In the above embodiment, although the sound source direction detecting means 20 utilizes the
microphones of the sound collecting means 10, it may be provided separately, or two or more
microphones may be used. It may be a method of spatially searching, or a method of utilizing
diffraction information of sound by placing an obstacle between two microphones like a human
head.
[0024]
As described above, according to the present invention, the sound source direction detecting
means for detecting the sound source direction to be collected and the sound collecting means
for collecting the sound in the direction detected by the sound source direction detecting means.
And automatic gain adjustment means for automatically adjusting the gain so that the output
level becomes constant each time the directivity of the sound is switched, so that directivity can
always be directed to the direction of the speaker, It can suppress the noise of air conditioning
and the multi-talker noise that was heard by many people at once, so that the voice of the
speaker can be heard more clearly.
Furthermore, by performing gain compensation using AGC means, the user does not have to
correct the volume controller each time for the voice of a distant speaker or a speaker with a
small voice. The correction is also performed gently only when the speaker changes, so that the
voice distortion that occurs in the conventional method of compressing the dynamic range using
the compressor does not occur, and the user operates the volume controller appropriately.
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Motion can be realized, and clear voices without distortion can always be heard, and even people
who have difficulty or can not hear voices in a conference can easily enjoy the other person's
comfort in the conference hall etc. It becomes possible to listen to people's stories.
[0025]
In addition, since the microphone array is used as the sound collection means, there is an
advantage that the directivity can be easily controlled, and the sound source direction can be
detected by using the microphones at both ends.
[0026]
Furthermore, since the delay-and-sum microphone array is used as the microphone array, there
is an advantage that the directivity is sharp and comfortable listening can be performed.
[0027]
Brief description of the drawings
[0028]
1 is a block diagram showing a basic configuration of an embodiment of the present invention.
[0029]
2 is a block diagram showing the configuration of an embodiment of the present invention.
[0030]
3 is a diagram for explaining the operation principle of the delay-and-sum microphone array
used in the embodiment of FIG.
[0031]
4 is an explanatory view showing an example of use of the present invention.
[0032]
Explanation of sign
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[0033]
DESCRIPTION OF SYMBOLS 10 sound collection means 11 microphone 12 microphone array 13
amplifier 14 delay sum circuit 15 variable delay element 16 adder 20 sound source direction
detection means 30 AGC means 40 amplification means 41 amplifier 42 gain multiplier 50
headphone
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