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DESCRIPTION JP2005057450

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DESCRIPTION JP2005057450
PROBLEM TO BE SOLVED: To adjust acoustic coupling between a speaker and a microphone
equally in a speech apparatus automatically and digitally using the equipment of a speech
apparatus without using a special apparatus. SOLUTION: The microphones MC1 to MC6
arranged radially are located equidistant from the speaker 16. A test signal such as a pink noise
signal is output from the test signal generation unit 258 in the digital signal processor (DSP) 25
to the speaker 16, and the microphone signal from which the sound of the speaker 16 has been
detected is A / D converter 271-with digital gain. At 273, the signal is input to the DSP 25 and
gain adjustment processing is performed by the level determination / gain control unit 257.
Based on the result, digital gain in the A / D converters 271 to 273, digital attenuation amount in
the variable attenuation unit 251 , The gain of the level output to the output amplifier 291, etc.
are adjusted. [Selected figure] Figure 32
Microphone-Speaker integrated configuration type-Communication device
[0001]
The present invention relates to, for example, a microphone-speaker integrated configurationcommunication device suitable for use when a plurality of conference participants in two
conference rooms conduct a conference by voice. In particular, in the present invention, in the
microphone / speaker integrated configuration / talking device, without using a special signal
generating device or signal measuring device, a speaker utilizing the function of the microphone
/ speaker integrated configuration / translation device itself is used. The present invention
relates to a technique for automatically adjusting the degree of acoustic coupling between a
microphone and a plurality of microphones.
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[0002]
A teleconferencing system is used in order for conference participants in two conference rooms
located at remote locations to conduct a conference. In the video conference system, the image of
the conference participants in each conference room is imaged by the imaging means, the sound
is collected by the microphone, and the image imaged by the imaging means and the sound
collected by the microphone are transmitted along the communication path. The image is
transmitted, and the picked up image is displayed on the display unit of the television receiver of
the conference room on the other party side, and the sound collected from the speaker is output.
[0003]
In such a video conference system, in each conference room, we have encountered the problem
that it is difficult for the voice of the speaker at a distance from the imaging means and the
microphone to be collected. A microphone may be provided for each participant. In addition,
there is also a problem that it is difficult for a participant in a conference located at a position
away from the speaker to hear the sound output from the speaker of the television receiver.
[0004]
Japanese Patent Application Laid-Open Nos. 2003-87887 and 2003-87890 are directed to the
other party in addition to the usual video conference system for providing video and audio when
carrying out a video conference in the mutually separated conference rooms. The microphone
and the speaker are integrally configured, which has the advantage that the voice of the meeting
attendee who is in the conference room can be clearly heard from the speaker and is not easily
affected by the noise in the conference room on this side or the burden of the echo canceler is
small. A voice input / output device is disclosed.
[0005]
For example, the voice input / output device disclosed in Japanese Patent Application Laid-Open
No. 2003-87887 is described with reference to FIGS. 5 to 8, 9 and 23 of Japanese Patent
Application Laid-Open No. 2003-87887. A speaker box 5 with a built-in speaker 6 from the
bottom to the top, a conical reflector 4 for diffusing sound radially open upward, a sound
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shielding plate 3 and a support 8 A plurality of unidirectional microphones (four in FIG. 6 and
seven in FIG. 23 and six in FIG. 23) are arranged radially at equal angles in the horizontal plane.
The sound shielding plate 3 is for shielding so that the sound from the lower speaker 5 does not
enter the plurality of microphones. JP 2003-87887 JP JP 2003-87890 JP
[0006]
The audio input / output devices disclosed in Japanese Patent Application Laid-Open Nos. 200387887 and 2003-87890 are utilized as means for complementing a video conference system
that provides video and audio. However, as the teleconferencing system, it is often sufficient to
use only the voice instead of using a complex device such as a video conference system. For
example, in the case where a plurality of conference participants hold meetings between the head
office of the same company and a remote sales office, they are also acquainted with each other
and understand each other's voice, so talk on the phone In addition, it is possible to hold a
conference sufficiently without video by the video conference system. Furthermore, introducing a
video conference system has the disadvantages that the amount of investment required to
introduce the video conference system itself, the complexity of operation, and the large
communication burden for transmitting captured images.
[0007]
Assuming that the present invention is applied to such an audio-only conference, in the audio
input / output devices disclosed in Japanese Patent Application Laid-Open Nos. 2003-87887 and
2003-87890, performance, price, and dimensional aspects are considered. And, in terms of
compatibility with the use environment, usability, etc., it is often improved.
[0008]
An object of the present invention is to provide a telephone apparatus further improved in terms
of performance, price, size, adaptability to a use environment, usability, etc. as means used only
for telephone calls.
In particular, according to the present invention, the acoustic coupling degree between the
speaker and the plurality of microphones is automatically made by utilizing the function of the
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microphone / speaker integrated type / communication device itself without using a special
signal generating device or signal measuring device. It is an object of the present invention to
provide a microphone / speaker integrated configuration / talking device which is made
adjustable.
[0009]
According to a first aspect of the present invention, a speaker, a plurality of microphones
equidistantly spaced from the speaker, an external output terminal, an input terminal, and a
function capable of digitally adjusting the gain are provided. A / D conversion means for
converting a detection signal of a microphone into a digital signal, level detection means for
digitally detecting the level of the conversion signal of the A / D conversion means, and signals
detected by the first level detection means Among them, signal selection means for selecting one
and outputting it to the external output terminal and the speaker, level determination / gain
control means for performing level determination and gain adjustment, and a digital test signal
are generated, and Echo cancellation is performed digitally on test signal generating means for
output, the signal output to the external output terminal, and the signal input from the input
terminal Echo cancellation means for performing processing, and the level determination and
gain control means may cause the detection signals of the plurality of microphones to have the
same degree of acoustic coupling between the speaker and the plurality of microphones. A
microphone-speaker integrated configuration / talking device is provided, characterized in that
the gain of the conversion means is digitally adjusted.
[0010]
Preferably, the apparatus further comprises second level detection means for detecting the level
of the signal output to the external output terminal, and the second level determination / gain
control means receives the signal detected by the second level detection means. To monitor and
adjust the level of the signal output to the external output terminal such that the level of the
signal output to the second external output terminal becomes a predetermined level.
[0011]
Preferably, the apparatus further comprises variable attenuation means capable of digitally
adjusting the amplitude of the signal converted by the A / D conversion means, the level
determination / gain control means comprising: a gain of the A / D conversion means; And / or
digitally adjust the amount of attenuation of the variable attenuation means.
[0012]
Specifically, the A / D conversion means converts every two microphone detection signals and
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has a gain common to the two signals, and the level determination / gain control means
calculates two microphone detection signals. The amount of attenuation of the variable
attenuation means is adjusted so that the sensitivity difference falls within a predetermined
range, and then one pair of two microphone detection signals is within a predetermined range of
the level of the other pair of two microphone detection signals. Thus, the gain of the A / D
conversion means is adjusted.
[0013]
Preferably, the echo cancellation means is an echo cancellation transmission process means
located between the signal selection means and the external output terminal and performing an
echo cancellation process on a transmission signal output to the speaker, and the input Echo
cancellation receiver means connected to a terminal for performing an echo cancellation process
digitally on a signal inputted from the outside, and the signal outputted to the external output
terminal is inputted to the echo cancellation receiver means.
[0014]
According to a second aspect of the present invention, there is provided a speaker, a plurality of
microphones disposed at an equal distance from the speaker, an external output terminal, an
input terminal, and an A / A converting a detection signal of the microphone into a digital signal.
D conversion means, variable attenuation means capable of digitally adjusting the amplitude of
the signal converted by the A / D conversion means, and first level detection means for digitally
detecting the level of the output signal of the variable attenuation means And signal selection
means for selecting one of the signals detected by the first level detection means and outputting
the selected signal to the external output terminal and the speaker; level determination / gain
control means for performing level determination and gain adjustment; Test signal generating
means for generating a digital test signal and outputting it to the input terminal, a signal output
to the external output terminal, and an input from the input terminal Echo cancellation means for
digitally performing an echo cancellation process on the input signal, and the level determination
/ gain control means detects that the detection signals of the plurality of microphones have the
degree of acoustic coupling between the speaker and the plurality of microphones. There is
provided a microphone-speaker integrated type communication device characterized by digitally
adjusting the attenuation amount of the variable attenuation means so as to be equal.
[0015]
Preferably, the apparatus further comprises second level detection means for detecting the level
of the signal output to the external output terminal, and the second level determination / gain
control means receives the signal detected by the second level detection means. To monitor and
adjust the level of the signal output to the external output terminal such that the level of the
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signal output to the second external output terminal becomes a predetermined level.
[0016]
In the present invention, a microphone, a speaker and a plurality of microphones are prepared
without preparing a special device by utilizing the function of the microphone and speaker
integrated configuration and the talk device used as the talk device. Acoustic coupling can be
automatically and easily equalized.
[0017]
First, an application example of the microphone / speaker integrated configuration type
communication device (hereinafter referred to as a communication device) of the present
invention will be described.
FIGS. 1A to 1C are configuration diagrams showing an example to which a microphone-speaker
integrated configuration type / communication device of the present invention is applied.
As illustrated in FIG. 1 (A), the telephones 1A and 1B are respectively installed in two remotely
located conference rooms 901 and 902, and the telephones 1A and 1B are connected by a
telephone line 920. There is.
As illustrated in FIG. 1B, in the two conference rooms 901 and 902, the communication devices
1A and 1B are placed on the tables 911 and 912, respectively.
However, in FIG. 1B, only the communication device 1A in the conference room 901 is illustrated
for simplification of the illustration.
The same applies to the communication device 1B in the meeting room 902.
An external perspective view of the call devices 1A and 1B is shown in FIG.
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As illustrated in FIG. 1C, a plurality of (six people in the present embodiment) conference
participants A1 to A6 are located around the communication devices 1A and 1B.
However, in FIG. 1 (C), only the conference participants around the communication device 1A in
the conference room 901 are illustrated in order to simplify the illustration. The arrangement of
the conference participants located around the communication device 1B in the other conference
room 902 is similar.
[0018]
The communication device of the present invention can respond, for example, by voice between
the two conference rooms 901 and 902 via the telephone line 920. In general, a conversation via
a telephone line 920 is a conversation between one speaker and one speaker, that is, one-to-one,
but the communication device of the present invention uses a single telephone line 920 to make
a plurality of conversations. The conference participants A1 to A6 can talk with each other.
However, although details will be described later, in order to avoid voice congestion, speakers at
the same time (in the same time zone) are mutually limited to one person. Since the
communication device of the present invention is intended for voice (call), it only transmits voice
via the telephone line 920. In other words, it does not transmit a large amount of image data as
in a video conference system. Furthermore, since the communication device of the present
invention compresses and transmits the call of the conference participant, the transmission load
on the telephone line 920 is light.
[0019]
Configuration of Communication Device The configuration of a communication device as one
embodiment of the present invention will be described with reference to FIGS. FIG. 2 is a
perspective view of the communication device as one embodiment of the present invention. FIG.
3 is a cross-sectional view of the communication device illustrated in FIG. FIG. 4 is a plan view of
the microphone / electronic circuit housing portion of the communication device illustrated in
FIG. 1, and is a plan view taken along the line XX-Y in FIG.
[0020]
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As illustrated in FIG. 2, the communication device 1 includes an upper cover 11, a sound
reflection plate 12, a connecting member 13, a speaker housing portion 14, and an operation
portion 15. As illustrated in FIG. 3, the speaker housing portion 14 has a sound reflecting surface
14 a, a bottom surface 14 b, and an upper sound output opening 14 c. The receiving and
reproducing speaker 16 is accommodated in a lumen 14 d which is a space surrounded by the
sound reflecting surface 14 a and the bottom surface 14 b. The sound reflection plate 12 is
located above the speaker housing portion 14, and the speaker housing portion 14 and the
sound reflection plate 12 are connected by the connection member 13.
[0021]
The restraining member 17 penetrates into the connecting member 13, and the restraining
member 17 is disposed between the restraining member lower fixing portion 14 e of the bottom
surface 14 b of the speaker housing portion 14 and the restraining member fixing portion 12 b
of the sound reflecting plate 12. Restraint. However, the restraint member 17 only penetrates the
restraint member penetrating portion 14 f of the speaker housing portion 14. Although the
restraining member 17 penetrates the restraining member penetrating portion 14f and is not
restrained here, the operation of the speaker 16 causes the speaker housing portion 14 to
vibrate, but the vibration is not restrained around the upper sound output opening 14c. It is for.
[0022]
Speaker The voice of the speaker in the conference room of the other party leaves the upper
sound output opening 14 c through the reception / playback speaker 16, and the sound
reflection surface 12 a of the sound reflection plate 12 and the sound reflection surface 14 a of
the speaker housing 14 It spreads in all directions 360 degrees around axis C-C along the defined
space. The cross section of the sound reflection surface 12a of the sound reflection plate 12
draws a gentle trumpet-shaped arc as illustrated. The cross section of the sound reflection
surface 12a has an illustrated cross-sectional shape over 360 degrees (all directions) around the
axis C-C. Similarly, the cross section of the sound reflecting surface 14a of the speaker housing
portion 14 also draws a gentle convex surface as illustrated. The cross section of the sound
reflection surface 14a also has the illustrated cross-sectional shape over 360 degrees (entire
orientation) about the axis C-C.
[0023]
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The sound S emitted from the reception / playback speaker 16 passes through the upper sound
output opening 14c, passes through a sound output space having a trumpet shape defined by the
sound reflection surface 12a and the sound reflection surface 14a, and the call apparatus 1 is
loaded. It spreads in 360 degrees all directions centered on the axis C-C along the surface of the
placed table 911 and is heard at a volume equal to all the conference participants A1 to A6. In
the present embodiment, the surface of the table 911 is also used as part of the sound
propagation means. The spread state of the sound S output from the reception and reproduction
speaker 16 is illustrated by an arrow.
[0024]
The sound reflection plate 12 supports the printed circuit board 21. The microphones MC1 to
MC6 of the microphone / electronic circuit housing portion 2, the light emitting diodes LED1 to
LED6, the microprocessor 23, the codec (CODEC) 24, and the first digital signal on the printed
circuit board 21 as illustrated in the plan view of FIG. Various electronic circuits such as a
processor (DSP) 25, a second digital signal processor (DSP) 26, an A / D converter block 27, a D /
A converter block 28, an amplifier block 29, etc. are mounted. Reference numeral 12 also
functions as a member for supporting the microphone / electronic circuit housing portion 2.
[0025]
The printed circuit board 21 has a damper 18 for absorbing the vibration from the reception /
playback speaker 16 so that the vibration from the reception / playback speaker 16 is
transmitted to the sound reflection plate 12 and enters the microphones MC1 to MC6 and the
like to avoid noise. It is attached. The damper 18 is composed of a screw and a buffer material
such as anti-vibration rubber inserted between the screw and the printed circuit board 21. The
buffer material is screwed to the printed circuit board 21 with the screw. That is, the vibration
transmitted from the reception / playback speaker 16 to the printed circuit board 21 is absorbed
by the buffer material. Thereby, the microphones MC1 to MC6 are less affected by the sound
from the speaker 16.
[0026]
Arrangement of Microphones As illustrated in FIG. 4, six microphones MC1 to MC6 are
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positioned at equal angles (at intervals of 60 degrees in this embodiment) radially from the
central axis C of the printed circuit board 21. Each microphone is a microphone with
unidirectionality. The characteristics will be described later. Each of the microphones MC1 to
MC6 is swingably supported by the first microphone support member 22a and the second
microphone support member 22b both having flexibility or elasticity (in order to simplify the
illustration, the microphones) The first and second microphone support members 22a and 22b in
the MC1 portion are illustrated only), in addition to the measures that are not affected by the
vibration from the reception reproduction speaker 16 by the damper 18 using the abovementioned cushioning material. , And the first and second microphone support members 22a and
22b having flexibility or elasticity absorb the vibration of the printed circuit board 21 vibrated by
the vibration from the reception and reproduction speaker 16, and the influence of the vibration
of the reception and reproduction speaker 16 The noise of the reception and reproduction
speaker 16 is avoided by not receiving the signal.
[0027]
As illustrated in FIG. 3, the receiving and reproducing speaker 16 is directed perpendicularly to
the central axis C-C of the plane on which the microphones MC1 to MC6 are located (in the
present embodiment, it is directed upward ( (Oriented), the arrangement of the reception and
reproduction speaker 16 and the six microphones MC1 to MC6 makes the distances between the
reception and reproduction speaker 16 and the microphones MC1 to MC6 equal. The sound
reaches the microphones MC1 to MC6 with almost the same volume and the same phase.
However, due to the configuration of the sound reflection surface 12a of the sound reflection
plate 12 and the sound reflection surface 14a of the speaker housing 14, the sound of the
reception / playback speaker 16 is not directly input to the microphones MC1 to MC6. In
addition, as described above, the influence of the vibration of the reception and reproduction
speaker 16 by using the damper 18 using the shock absorbing material and the first and second
microphone support members 22a and 22b having flexibility or elasticity. Is reduced. As
illustrated in FIG. 1C, the conference participants A1 to A6 generally have substantially equal
intervals in the vicinity of the microphones MC1 to MC6 disposed at an interval of 60 degrees in
the direction of 360 degrees around the communication device 1 Located at.
[0028]
Light-emitting diodes Light-emitting diodes LED1 to LED6 are arranged in the vicinity of the
microphones MC1 to MC6 as an example of means (microphone selection result display means
30) for notifying that the speaker has been determined, which will be described later. The light
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emitting diodes LED1 to LED6 are provided so as to be visible from all the conference
participants A1 to A6 even when the upper cover 11 is attached. Therefore, the upper cover 11 is
provided with a transparent window so that the light emission state of the light emitting diodes
LED1 to 6 can be visually recognized. Of course, an opening may be provided in the upper cover
11 in the light emitting diodes LED1 to LED6, but a light transmitting window is preferable from
the viewpoint of dust protection to the microphone / electronic circuit housing portion 2.
[0029]
A first digital signal processor (DSP) 25, a second digital signal processor (DSP) 26, and various
electronic circuits 27 to 29 perform various kinds of signal processing to be described later on
the printed circuit board 21. It is arrange | positioned in space other than the part in which MC6
is located. In the present embodiment, the DSP 25 is used as signal processing means for
performing processing such as filter processing and microphone selection processing together
with various electronic circuits 27 to 29, and the DSP 26 is used as an echo canceler.
[0030]
FIG. 5 is a schematic configuration diagram of the microprocessor 23, the codec 24, the DSP 25,
the DSP 26, the A / D converter block 27, the D / A converter block 28, the amplifier block 29,
and various other electronic circuits. The microprocessor 23 performs overall control processing
of the microphone / electronic circuit housing unit 2. The codec 24 compresses and encodes the
voice to be transmitted to the other party's conference room. The DSP 25 performs various types
of signal processing described below, such as filtering and microphone selection. The DSP 26
functions as an echo canceller, and has an echo cancellation transmission processing unit 261
and an echo cancellation reception unit 262. In FIG. 5, four A / D converters 271 to 274 are
illustrated as an example of the A / D converter block 27, and two D / A converters as an
example of the D / A converter block 28. 281 and 282 are illustrated, and two amplifiers 291
and 292 are illustrated as an example of the amplifier block 29. In addition, various circuits such
as a power supply circuit are mounted on the printed circuit board 21 as the microphone /
electronic circuit housing unit 2.
[0031]
In FIG. 4, a pair of microphones MC1-MC4: MC2-MC5: MC3-M6 disposed on a straight line at
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respective symmetrical (or opposed) positions with respect to the central axis C of the printed
circuit board 21 respectively have two channels. It is input to A / D converters 271 to 273 which
convert analog signals to digital signals. In the present embodiment, one A / D converter converts
2-channel analog input signals into digital signals. Therefore, the detection signals of two (one
pair) microphones, for example, microphones MC1 and MC4 located on a straight line
sandwiching central axis C, are input to one A / D converter and converted into digital signals.
There is. Further, in the present embodiment, in order to identify the speaker of the voice to be
sent to the conference room of the other party, the difference between the voices of the two
microphones positioned on a straight line, the magnitude of the voice, etc. are referred to. When
the signals of two microphones located on the line are input to the same A / D converter, the
conversion timing is almost the same, and when the difference between the audio outputs of the
two microphones is taken, the timing error is small. There are advantages such as ease. The A / D
converters 271 to 274 can also be configured as A / D converters 271 to 274 with a variable
gain amplification function. The sound collection signals of the microphones MC1 to MC6
converted by the A / D converters 271 to 273 are input to the DSP 25, and various signal
processing described later is performed. As one of the processing results of the DSP 25, the
result of selecting one of the microphones MC1 to MC6 is output to the corresponding one of the
light emitting diodes LED1 to LED6 which is an example of the microphone selection result
display means 30.
[0032]
The processing result of the DSP 25 is output to the DSP 26 and echo cancellation processing is
performed. The DSP 26 has, for example, an echo cancellation transmission processing unit 261
and an echo cancellation reception unit 262. The processing result of the DSP 26 is converted to
an analog signal by the D / A converters 281 and 282. The output from the D / A converter 281
is optionally encoded by the codec 24 and output to the line out of the telephone line 920 (FIG.
1A) through the amplifier 291 to the other party's conference room It is outputted as a sound
through the reception reproduction speaker 16 of the installed telephone apparatus 1. The voice
from the communication apparatus 1 installed in the conference room of the other party is input
through the line-in of the telephone line 920 (FIG. 1A), converted into a digital signal by the A / D
converter 274, and converted to the DSP 26. It is input and used for echo cancellation
processing. Further, the sound from the telephone apparatus 1 installed in the conference room
of the other party is applied to the speaker 16 through a path not shown and is output as a
sound. The output from the D / A converter 282 is output as a sound from the reception and
reproduction speaker 16 of the communication device 1 through the amplifier 292. That is, in
addition to the voice of the speaker selected in the other party's conference room from the
above-mentioned reception and playback speaker 16, the conference participants A1 to A6 also
receive the voice of the speaker in the conference room. You can listen through.
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[0033]
Microphones MC1 to MC6 FIG. 6 is a graph showing the characteristics of the microphones MC1
to MC6. The frequency characteristics and level characteristics of each single directional
microphone change as shown in FIG. 6 according to the arrival angle of voice from the speaker to
the microphone. The plurality of curves show the directivity when the frequency of the collected
signal is 100 Hz, 150 Hz, 200 Hz, 300 Hz, 400 Hz, 500 Hz, 700 Hz, 1000 Hz, 1500 Hz, 2000 Hz,
3000 Hz, 4000 Hz, 5000 Hz, 7000 Hz. However, to simplify the illustration, FIG. 6 typically
illustrates the directivity for 150 Hz, 500 Hz, 1500 Hz, 3000 Hz, and 7000 Hz.
[0034]
FIGS. 7A to 7D are graphs showing the results of spectrum analysis of the position of the sound
source and the sound collection level of the microphone, and as an example of analysis, a
distance of 1.5 meters, for example, 1.5 meters from the communication device 1 It shows the
results of fast Fourier transform (FFT) of voices collected by microphones with a speaker at a
distance at fixed time intervals. The X axis represents frequency, the Y axis represents signal
level, and the Z axis represents time. When the directional microphone shown in FIG. 6 is used,
strong directivity is shown in front of the microphone. In the present embodiment, the DSP 25
performs microphone selection processing utilizing such characteristics.
[0035]
When a nondirectional microphone is used instead of a directional microphone as in the present
invention, all sound around the microphone is collected, so the S / N of the speaker's voice and
the peripheral noise are mixed up too much. I can not collect good sounds. In order to avoid this,
in the present invention, the S / N with the surrounding noise is improved by collecting sound
with one directional microphone. A microphone array using multiple omnidirectional
microphones can be used as a method of obtaining the directivity of the microphone, but such a
method requires complicated processing due to the coincidence of time axes (phases) of multiple
signals. Time and response are low, and the device configuration is complicated. That is, the
signal processing system of the DSP also requires complicated signal processing. The present
invention solves such a problem using the directional microphone illustrated in FIG. In order to
combine the microphone array signal and use it as a directional sound collecting microphone,
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there is a disadvantage that the outer shape is restricted by the pass frequency characteristic and
the outer shape becomes large. The present invention also solves this problem.
[0036]
Effects of Device Configuration of Communication Device The communication device having the
above-described configuration exhibits the following advantages. (1) The positional relationship
between the even number microphones MC1 to MC6 arranged at equal angles radially and at
equal intervals and the reception / reproduction speaker 16 is constant, and further the distance
is very short. The level of the sound coming back directly from the level coming back to the
microphones MC1 to MC6 through the conference room (room) environment is overwhelmingly
dominant. Therefore, the characteristics (signal level (intensity), frequency characteristics (f
characteristic), and phase) of the sound reaching the microphones MC1 to MC6 from the speaker
16 are always the same. That is, in the communication device 1 in the embodiment of the present
invention, there is an advantage that the transfer function is always the same. (2) Therefore,
there is no change in transfer function when the output of the microphone to be sent to the other
party's conference room is switched when the speakers are different, and it is not necessary to
adjust the gain of the microphone system each time the microphone is switched. Have an
advantage. In other words, there is an advantage that once the adjustment is made at the time of
manufacturing the communication device, there is no need to perform the adjustment again. (3)
Even if the microphones are switched when the speakers are different for the same reason as
described above, only one echo canceller (DSP 26) may be used. The DSP is expensive, there is no
need to arrange a plurality of DSPs on the printed circuit board 21 with various members
mounted thereon and space is small, and the space for arranging the DSPs on the printed circuit
board 21 may be small. As a result, the printed circuit board 21 and hence the communication
device of the present invention can be miniaturized. (4) As described above, since the transfer
function between the reception and reproduction speaker 16 and the microphones MC1 to MC6
is constant, for example, there is an advantage that the sensitivity difference adjustment of the
microphone itself having ▒ 3 dB can be performed by the microphone unit of the
communication device alone. There is. Details of sensitivity difference adjustment will be
described later. (5) The table on which the communication device 1 is mounted usually uses a
round table (round table) or a polygonal table, but one reception / playback speaker 16 in the
communication device 1 centers voice of uniform quality around the axis C. As the speaker
system becomes possible to disperse evenly (dispersion) to 360 degrees omnidirectional. (6) The
sound from the reception / playback speaker 16 is transmitted to the table surface of the round
table (boundary effect), and the quality sound effectively and evenly reaches the conference
participants, and the ceiling direction of the conference room is opposed The side sound and
phase are canceled to make a small sound, and there is an advantage that the reflected sound
from the ceiling direction to the conference participants is small, and as a result, a clear sound is
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distributed to the participants.
(7) The sound coming from the reception / playback speaker 16 reaches the microphones MC1
to MC6 arranged at equal angles radially and at equal intervals at the same volume at the same
time, so it is judged whether it is the speaker's voice or the reception voice. It will be easier. As a
result, erroneous determination of the microphone selection process is reduced. The details will
be described later. (8) An even number of microphones, for example, six microphones arranged
at equal angles radially and at equal intervals, and a pair of opposing microphones arranged on a
straight line, for example, a level comparison for detecting the direction of a sound source Can be
done easily. (9) The damper 18, the microphone support member 22 and the like can reduce the
influence of the vibration due to the sound of the reception / playback speaker 16 on the sound
collection of the microphones MC <b> 1 to MC <b> 6. (10) As illustrated in FIG. 3, structurally,
the sound of the reception / playback speaker 16 is less directly propagated to the microphones
MC1 to MC6. Therefore, in the communication apparatus 1, the influence of noise from the
reception and reproduction speaker 16 is small.
[0037]
Modifications Although the telephone set 1 described with reference to FIGS. 2 to 3 arranges the
receiving and reproducing speaker 16 in the lower part, and arranges the microphones MC1 to
MC6 (and related electronic circuits) in the upper part, the receiving and reproducing speaker
The positions of the microphone 16 and the microphones MC1 to MC6 (and the associated
electronics) can also be upside down, as illustrated in FIG. Even in such a case, the abovedescribed effect is obtained.
[0038]
The number of microphones is not limited to six, and an arbitrary even number of four
microphones, eight microphones, etc. are equally spaced radially at equal angles, equidistantly
spaced from each other on an axis C, in parallel (same direction) , And microphones MC1 and
MC4 are arranged in a straight line. The reason for arranging two microphones, for example,
MC1 and MC4 to face each other in a straight line is to easily and accurately identify a speaker.
[0039]
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15
Content of Signal Processing The content of processing performed by the first digital signal
processor (DSP) 25 will be mainly described below. FIG. 9 is a diagram illustrating an outline of
processing performed by the DSP 25. As shown in FIG. The outline is described below.
[0040]
(1) Measurement of ambient noise As an initial operation, preferably, ambient noise in which the
two-way communication device 1 is installed is measured. The communication device 1 can be
used in various environments (conference rooms). In order to ensure the accuracy of the
selection of the microphone and to improve the performance of the communication device 1, in
the present invention, the noise of the surrounding environment in which the communication
device 1 is installed is measured at an early stage, and the effect of the noise is measured by the
microphone It is possible to exclude it from the collected signal. Of course, when the calling
device 1 is used repeatedly in the same conference room, noise measurement is performed in
advance, and this processing can be omitted if the noise condition does not change. Note that
noise measurement can also be performed in a normal state. Details of the noise measurement
will be described later.
[0041]
(2) Selection of Chairperson For example, when using the telephone apparatus 1 for a two-way
conference, it is useful that there is a chairperson who organizes the management of the
proceedings in each conference room. Therefore, as an aspect of the present invention, a
chairperson is set from the operation unit 15 of the speech device 1 at the initial stage of using
the speech device 1. As a setting method of the chairperson, for example, the first microphone
MC1 located in the vicinity of the operation unit 15 is used as the chairperson microphone. Of
course, the chairman microphone can also be optional. In addition, this process can be omitted,
when the chairperson who uses the telephone apparatus 1 repeatedly is the same. Alternatively,
the microphone in which the chairperson sits may be determined in advance. In that case, the
selection operation of the chairman is not necessary each time. Of course, the selection of the
chairman is not limited to the initial state, but can be performed at any timing. Details of the
chairperson selection will be described later.
[0042]
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16
(3) Adjustment of sensitivity difference of microphones As an initial operation, preferably, a gain
or attenuation unit of an amplification unit that amplifies signals of microphones MC1 to MC6 so
that acoustic coupling between reception / reproduction speaker 16 and microphones MC1 to
MC6 becomes equal. Automatically adjust the attenuation value of. The sensitivity difference
adjustment will be described later.
[0043]
As the normal processing, various types of processing exemplified below are performed. (3)
Microphone selection and switching processing When a plurality of conference participants talk
at the same time in one conference room, the voice is mixed and it is difficult for the conference
participants A1 to A6 in the other conference room to hear. Therefore, in the present invention,
in principle, one person talks at a certain time zone. Therefore, the DSP 25 performs speaker
identification and selection / switching processing of a microphone for allowing a call. As a
result, only the call from the selected microphone is transmitted to the communication apparatus
1 of the other party's conference room via the telephone line 920 and output from the speaker.
Of course, as described with reference to FIG. 5, the LED in the vicinity of the microphone of the
selected speaker is lit, and furthermore, the voice of the selected speaker is also heard from the
speaker of the communication device 1 in the room. Can identify who is the authorized speaker.
By this processing, it is possible to select the signal of the unidirectional microphone facing the
speaker and send a good S / N signal to the other party as the transmission signal. (4) Display of
Selected Microphone A microphone is selected so that all the conference participants A1 to A6
can easily recognize which microphone of the conference participant who is permitted to speak
and who is the speaker microphone is selected. The selection result display means 30, for
example, the corresponding ones of the light emitting diodes LED1 to 6 are lit. (5) As background
art of the above-mentioned microphone selection processing, or in order to execute the
microphone selection processing correctly, various kinds of signal processing exemplified below
are performed. (A) Band separation of the sound collection signal of the microphone and level
conversion processing (b) Judgment processing of start and end of speech In order to use as a
selection judgment start trigger of the microphone signal facing in the direction of the speaker.
(C) Detection processing of microphones in the direction of the speaker In order to analyze the
sound collection signal of each microphone and to determine the microphone used by the
speaker. (D) Speaker-direction microphone switching timing determination processing, and
selection switching processing of microphone signal facing the detected speaker The switching
instruction is given to the microphone selected from the processing result described above. (E)
Measurement of floor noise during normal operation
10-04-2019
17
[0044]
Floor (environment) noise measurement This process is divided into an initial process and a
normal process immediately after the two-way communication device is powered on. In addition,
this process is performed under the following exemplary preconditions.
[0045]
[Table 1] (1) Condition: Measurement time and threshold provisional value: 1. Test tone sound
pressure: -40 dB at microphone signal level Noise measurement unit time: 10 seconds 3. Noise
measurement under normal conditions: Calculate the average value from the measurement
results for 10 seconds, and repeat this 10 times to obtain the average value and use it as the
noise level.
[0046]
[Table 2] (2) Estimated effective distance and threshold by difference between floor noise and
speech start reference level 1.26 dB or more: 3 meters or more Speech start detection level
threshold: floor noise level + 9 dB Speech end detection level threshold: Floor noise level + 6 dB
2. 20-26 dB: Within 3 meters Detection level threshold for speech start: Floor noise level + 9 dB
Detection level threshold for speech end: Floor noise level + 6 dB 3. 14-20 dB: Within 1.5 meters
Detection of speech start Level threshold: Floor noise level + 9 dB Detection level threshold for
speech end: Floor noise level + 6 dB 4.9 to 14 dB: Within 1 meter Detection level threshold for
speech start: Difference between floor noise level and speech start reference level э 2 + 2 dB
Detection level threshold: speech start threshold-3 dB 9 dB or less: a little tight, several tens of
centimeters Detection level threshold of speech start: 6. Difference between floor noise level and
speech start reference level 2 Detection level threshold for speech end: -3 dB 7. Same or negative:
can not be judged and selection prohibited
[0047]
[Table 3] (3) The noise measurement start threshold value of the normal process starts from the
time when the floor noise level at power on + 3 dB or less.
[0048]
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18
Immediately after the power of the communication device 1 is turned on, the DSP 25 performs
the following noise measurement described with reference to FIGS.
The initial processing of the DSP 25 immediately after the power supply to the speech apparatus
1 measures floor noise and reference signal level, and based on the difference, sets an indication
of effective distance between the speaker and the present system and setting of speech start and
end judgment threshold levels. To do. The level value peak-held by the sound pressure level
detection unit in the DSP 25 is read at a constant time interval, for example, 10 mSec, and the
average value of the unit time values is calculated to be floor noise. Then, the DSP 25 determines
the threshold of the speech start detection level and the speech end detection level based on the
measured floor noise level.
[0049]
10, process 1: test level measurement The DSP 25 outputs a test tone to the line-in terminal of
the reception signal system illustrated in FIG. 5 according to the process illustrated in FIG. 10,
and the sound from the reception reproduction speaker 16 is transmitted to each microphone
MC1. The sound is collected by MC6, and the signal is used as a speech start reference level to
obtain an average value.
[0050]
11, process 2: noise measurement 1 The DSP 25 collects the levels of the sound collection signals
from the microphones MC1 to MC6 as floor noise levels for a certain period of time according to
the process illustrated in FIG. 11, and obtains an average value.
[0051]
12, process 3: estimation of the effective distance The DSP 25 compares the speech start
reference level with the floor noise level according to the process illustrated in FIG. 12, and
estimates the noise level of a room such as a conference room where the communication
apparatus 1 is installed. Then, the communication device 1 calculates the effective distance
between the speaker working well and the communication device 1.
[0052]
Microphone selection prohibition determination As a result of processing 3, if the floor noise is
higher (higher) than the speech start reference level, the DSP 25 determines that there is a
10-04-2019
19
strong noise source in the direction of the microphone, and the microphone of that direction is
automatically The selection is set to be prohibited and displayed, for example, on the microphone
selection result display means 30 or the operation unit 15.
[0053]
Threshold Determination As illustrated in FIG. 13, the DSP 25 compares the speech start
reference level with the floor noise level, and determines the threshold of the speech start / end
level from the difference.
[0054]
As far as noise measurement is concerned, since the next processing is normal processing, the
DSP 25 sets each timer (counter) to prepare for the next processing.
[0055]
Normal noise processing The DSP 25 performs noise processing according to the processing
shown in FIG. 14 in the normal operation state even after the above-mentioned noise
measurement at the initial operation of the telephone set 1, and selects each of the six
microphones MC1 to MC6. The volume level average value of the speaker and the noise level
after detection of the speech end are measured, and the speech start and end judgment threshold
levels are reset in fixed time units.
[0056]
FIG. 14, Process 1: The DSP 25 determines a branch to Process 2 or Process 3 based on the
judgment as to whether the speech is in progress or speech completion.
[0057]
FIG. 14, Process 2: Speaker Level Measurement The DSP 25 averages the level data for a unit
time during speech, for example, 10 seconds a plurality of times, for example, 10 times, and
records it as a speaker level.
When the speech ends in the unit time, the time measurement and the speech level measurement
are stopped until the start of a new speech, and the measurement process is restarted after the
new speech is detected.
10-04-2019
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[0058]
14, process 3: floor noise measurement 2 The DSP 25 averages noise level data for a unit time,
for example, 10 seconds, from a speech end detection to a speech start for a plurality of times,
for example, 10 times to calculate a floor noise level. Record as
If there is a new message within the unit time, the DSP 25 stops time measurement and noise
measurement on the way, and restarts the measurement process after detecting the end of the
new message.
[0059]
14, processing 4: threshold determination 2 The DSP 25 compares the speech level and the floor
noise level, and determines the threshold of speech start and end levels from the difference.
In addition to this, since the average value of the speech level of the speaker is obtained as an
application, it is possible to set the speech start and end detection threshold levels specific to the
speaker facing the microphone.
[0060]
Generation of Various Frequency Component Signals by Filter Processing FIG. 15 is a
configuration diagram showing a filtering processing performed by the DSP 25 as a preprocessing of a sound signal collected by a microphone.
FIG. 15 shows processing for one microphone (channel (one sound collection signal)).
The collected sound signal of each microphone is processed by, for example, an analog low-cut
filter 101 having a cutoff frequency of 100 Hz, and a filtered audio signal from which a
frequency of 100 Hz or less has been removed is output to the A / D converter 102. Digital high-
10-04-2019
21
cut filters 103a to 103e (collectively referred to as having a cut-off frequency of 7.5 KHz, 4 KHz,
1.5 KHz, 600 Hz, 250 Hz, respectively) of the collected sound signals converted into digital
signals by the A / D converter 102. The high frequency component is removed at step 103) (high
cut process).
The results of the digital high cut filters 103a to 103e are further subjected to subtraction for
each filter signal of the adjacent digital high cut filters 103a to 103e in subtractors 104a to 104d
(collectively, 104).
In the embodiment of the present invention, the digital high cut filters 103a to 103e and the
subtractors 104a to 104d are actually processed in the DSP 25.
The A / D converter 102 can be implemented as one of the A / D converter blocks 27.
[0061]
FIG. 16 is a frequency characteristic diagram showing the filter processing result described with
reference to FIG. A plurality of signals having various frequency components are generated from
the signal collected by the microphone having one directivity in this manner.
[0062]
Band-pass filter processing and microphone signal level conversion processing One of the
triggers for start of microphone selection processing is determination of the start and end of
speech. A signal used for that purpose is obtained by the band pass filter processing and level
conversion processing illustrated in FIG. FIG. 17 shows only one CH during input signal
processing of six channels (CH) collected by the microphones MC1 to MC6. The band pass filter
processing and level conversion processing unit in the DSP 25 are 100 to 600 Hz, 200 to 250
Hz, 250 to 600 Hz, 600 to 1500 Hz, 1500 to 4000 Hz, 4000 to 7500 Hz, respectively. Band pass
filters 201a to 201a (collectively, band pass filters block 201) having band pass characteristics,
and level converters 202a to 202g (level converting the original microphone sound collection
signal and the band-pass sound collection signal) Collectively, the level conversion block 202) is
included.
10-04-2019
22
[0063]
Each of the level conversion units 202 a to 202 g has a signal absolute value processing unit 203
and a peak hold processing unit 204. Therefore, as exemplified in the waveform, the signal
absolute value processing unit 203 inverts the sign and converts it into a positive signal when
the negative signal indicated by the broken line is input. The peak hold processing unit 204 holds
the maximum value of the output signal of the signal absolute value processing unit 203.
However, in the present embodiment, the maximum value held decreases somewhat with the
passage of time. Of course, the peak hold processing unit 204 can be improved to reduce the
amount of decrease and hold the maximum value for a long time.
[0064]
Describe band pass filter. The band pass filter used for the communication device 1 is, for
example, a band pass filter composed of only a second-order IIR high cut filter and a low cut filter
of the microphone signal input stage. In the present embodiment, it is used that when the signal
having a high frequency pass filter is subtracted from the signal having a flat frequency
characteristic, the rest is almost equal to the signal having a low frequency cut filter. In order to
match frequency-level characteristics, one band extra band pass filter is required, but the number
of band pass filters required + number of filter stages + the number of filter stages and the
required band Pass is obtained. The band frequency of the hand pass filter required at this time
is the following six band band pass filters per channel (CH) of the microphone signal.
[0065]
[Table 4] BP characteristic band pass filter BPF1 = [100Hz-250Hz] и и и 201b BPF2 = [250Hz600Hz] и и 201c BPF3 = [600Hz-1.5KHz] и и 201d BPF4 = [1.5KHz-4KHz] и и и 201e BPF5 = [4 KHz7.5 KHz] и и 201 f BPF 6 = [100 Hz-600 Hz] и и 201 a
[0066]
In this method, the calculation program of the above-mentioned IIR filter in the DSP 25 is only
6CH (channel) О 5 (IIR filter) = 30.
10-04-2019
23
Contrast with the conventional band pass filter configuration. Assuming that the configuration of
the band-pass filter uses a second-order IIR filter, if six-band band-pass filters are prepared for
each of six microphone signals as in the present invention, in the conventional method, 6 О 6 О
2 = 72. The circuit needs IIR filtering. This process requires considerable program processing
even with the latest excellent DSPs and affects other processes. In the embodiment of the present
invention, the 100 Hz low cut filter is processed by the analog filter of the input stage. There are
five cutoff frequencies of 250 Hz, 600 Hz, 1.5 KHz, 4 KHz, and 7.5 KHz for the second-order IIR
high-cut filter to be prepared. Of these, the high cut filter with a cut-off frequency of 7.5 KHz is
not necessary because the sampling frequency is actually 16 KHz, but in the process of
subtraction processing, the output level of the band pass filter decreases due to the influence of
the phase around the IIR filter. In order to reduce the phenomenon, the phase of the reduced
number is intentionally rotated.
[0067]
FIG. 18 is a flowchart when processing by the configuration illustrated in FIG. 17 is processed by
the DSP 25.
[0068]
The filter processing in the DSP 25 illustrated in FIG. 18 performs high-pass filter processing as
processing of the first stage and subtraction processing from the result of high-pass filter
processing of the first stage as processing of the second stage.
FIG. 16 is an image frequency characteristic diagram of the signal processing result. The
following [x] shows each processing case in FIG.
[0069]
First stage [1] For the whole band pass filter, the input signal is passed through a 7.5 KHz high
cut filter. This filter output signal becomes a band pass filter output of [100 Hz-7.5 KHz] by
analog low cut matching of the input.
[0070]
10-04-2019
24
[2] Pass the input signal through a 4 KHz high cut filter. This filter output signal becomes a [100
Hz-4 KHz] de-pass filter output in combination with the input analog low cut filter.
[0071]
[3] The input signal is passed through a 1.5 KHz high cut filter. This filter output signal is
combined with the analog low-cut filter at the input [100Hz-1.5KHz] by the combination with the
analog low-cut filter at the input [100Hz-1.5KHz] by the combination with the analog low-cut
filter at the input [100Hz -1.5 KHz] bandpass filter output.
[0072]
[4] The input signal is passed through a 600 KHz high cut filter. This filter output signal becomes
a band pass filter output of [100 Hz-600 Hz] in combination with the analog low cut filter of the
input.
[0073]
[5] The input signal is passed through a 250 KHz high cut filter. This filter output signal becomes
a band pass filter output of [100 Hz-250 Hz] in combination with the analog low cut filter of the
input.
[0074]
Second stage [1] When the bandpass filter (BPF5 = [4 KHz to 7.5 KHz]) executes the processing
of the filter outputs [1]-[2] ([100 Hz to 7.5 KHz]-[100 Hz to 4 KHz]) The signal output [4 KHz to
7.5 KHz] is obtained. [2] The band pass filter (BPF4 = [1.5 KHz to 4 KHz]) executes the processing
of the filter output [2]-[3] ([100 Hz to 4 KHz]-[100 Hz to 1.5 KHz]), the above signal The output is
[1.5 KHz to 4 KHz]. [3] The band pass filter (BPF3 = [600 Hz to 1.5 KHz]) executes the processing
of the filter output [3]-[4] ([100 Hz to 1.5 KHz]-[100 Hz to 600 Hz]) The output is [600 Hz to 1.5
KHz]. [4] The band pass filter (BPF2 = [250 Hz to 600 Hz]) executes the processing of the filter
output [4]-[5] ([100 Hz to 600 Hz]-[100 Hz to 250 Hz]) and the above signal output [250 Hz ~
10-04-2019
25
600 Hz]. [5] The band pass filter (BPF1 = [100 Hz to 250 Hz]) uses the signal of [5] as it is as the
output signal [5]. [6] The band pass filter (BPF 6 = [100 Hz to 600 Hz]) uses the signal of [4] as it
is as the output signal of (4). The band pass filter output required by the above processing in the
DSP 25 is obtained.
[0075]
The input sound collection signals MIC1 to MIC6 of the microphones are constantly updated as
shown in Table 5 as the sound pressure level of the entire band and the sound pressure level of
the six bands passed through the band pass filter in the DSP 25.
[0076]
[0077]
In Table 5, for example, L1-1 indicates the peak level when the sound collection signal of the
microphone MC1 passes the first band pass filter 201a.
The speech start / end determination uses the microphone sound collection signal that has
passed through the 100 Hz to 600 Hz band pass filter 201a illustrated in FIG. 17 and whose
sound pressure level has been converted by the level conversion unit 202b.
[0078]
Since the conventional band pass filter configuration is performed by combining a high pass filter
and a low pass filter per band pass filter stage, 36 band pass filters based on the specifications
used in the present embodiment are provided. Would require filtering of 72 circuits.
On the other hand, the filter configuration of the embodiment of the present invention is
simplified as described above.
[0079]
10-04-2019
26
Speech start / end determination processing The first digital signal processor (DSP1) 25
generates a microphone noise collection signal level based on floor noise as illustrated in FIG. 19
based on the value output from the sound pressure level detection unit. Rising, when it exceeds
threshold of speech start level It decides that it is speech start, when level which is higher than
threshold of start level continues, during speech, when floor level becomes lower than threshold
of speech end, it decides that it is floor noise When the speech end determination time is
continued, for example, for 0.5 seconds, it is determined that the speech is ended. The sound
pressure level data (microphone signal level (1)) that has passed through the 100 Hz to 600 Hz
band pass filter whose sound pressure level has been converted by the microphone signal
conversion processing unit 202b illustrated in FIG. It is determined that the speech is started
when the threshold level illustrated in FIG. 19 or more is reached. The DSP 25 is configured not
to detect the next speech start for a speech end determination time, for example, 0.5 seconds,
after detecting the speech start, in order to avoid operation failure caused by frequent
microphone switching.
[0080]
Microphone Selection The DSP 25 performs the speaker direction detection in the mutual
communication system and the automatic selection of the microphone signal facing the speaker
based on a so-called "star table system". FIG. 20 is a graph illustrating the operation of the
communication device 1. FIG. 21 is a flowchart showing the normal processing of the
communication device 1.
[0081]
As illustrated in FIG. 20, the communication device 1 performs audio signal monitoring
processing according to the sound collection signals from the microphones MC1 to MC6,
performs speech start / end determination, performs speech direction determination, and
performs microphone selection. The result is displayed on the microphone selection result
display means 30, for example, the light emitting diodes LED1 to LED6. Hereinafter, the
operation will be described mainly with the DSP 25 in the communication device 1 with
reference to the flowchart in FIG. The overall control of the microphone / electronic circuit
housing unit 2 is performed by the microprocessor 23, but the processing of the DSP 25 will be
mainly described.
10-04-2019
27
[0082]
Step 1: Monitoring of Level Conversion Signal The signals collected by the microphones MC1 to
MC6 are respectively transmitted to the band pass filter block 201 and the level conversion block
202 described with reference to FIGS. Since the seven types of level data are converted, the DSP
25 constantly monitors seven types of signals for each microphone sound collection signal.
Based on the monitoring result, the DSP 25 shifts to any one of the speaker direction detection
process 1, the speaker direction detection process 2, and the speech start / end determination
process.
[0083]
Step 2: Speech start / end judgment processing The DSP 25 judges the start / end of the speech
according to the method described in detail below with reference to FIG. When the DSP 25
detects the speech start, it notifies the judgment process of the direction of the speaker of step 4
that the speech start is detected. In addition, when the speech level is smaller than the speech
end level, the process of judging the start and end of the speech in step 2 starts a timer for the
speech end judgment time (for example, 0.5 seconds) and the speech end judgment time and the
speech level are speech When it is lower than the end level, it is determined that the speech is
ended. If it becomes higher than the speech end level within the speech end determination time,
the process of waiting is made until it becomes smaller than the speech end level.
[0084]
Step 3: Speaker Direction Detection Process The speaker direction detection process in the DSP
25 is performed by continuously searching the speaker direction. Thereafter, the data is supplied
to the determination process of the speaker direction in step 4.
[0085]
Step 4: Speaker Direction Microphone Switching Processing The timing determination processing
in the processing for switching the speaker direction microphone to the DSP 25 has been
selected from the speaker detection direction at that time based on the processing in Step 2 and
the processing in Step 3. When the speaker direction is different, the microphone selection of the
10-04-2019
28
new speaker direction is instructed to the microphone signal switching process of step 4.
However, if the chairman's microphone is set from the operation unit 15 and the chairman's
microphone and other conference participants speak simultaneously, priority is given to the
chairman's speech. At this time, the selected microphone information is displayed on the
microphone selection result display means 30, for example, the light emitting diodes LED1 to
LED6.
[0086]
Step 5: Transmission of Microphone Sound Collection Signal In the microphone signal switching
processing, only the microphone signal selected by the processing in step 4 out of the six
microphone signals is used as a transmission signal, and the other party from the telephone
apparatus 1 through the telephone line 920 To the line unit of the telephone line 920 illustrated
in FIG.
[0087]
Setting of speech start level threshold and speech end threshold Process 1: Measure floor noise
for a predetermined time, for example, one second, of each microphone immediately after power
on.
The DSP 25 reads out the peak-held level value of the sound pressure level detection unit at
constant time intervals, for example, 10 mSec intervals in this embodiment, and calculates an
average value of values for a predetermined time, for example, 1 minute, and floor noise I
assume. The DSP 25 determines the speech start detection level (floor noise +9 dB) and the
speech end detection level threshold (floor noise +6 dB) based on the measured floor noise level.
The DSP 25 subsequently reads out the peak-held level value of the sound pressure level detector
at constant time intervals. When it is determined that the speech is ended, the DSP 25 works as a
measurement of floor noise, detects the start of speech, and updates the threshold of the
detection level of the speech end.
[0088]
According to this method, since the threshold setting is different for each floor noise level at the
position where the microphone is placed, the threshold can be set for each microphone, and
erroneous determination in selection of the microphone by the noise source can be prevented.
10-04-2019
29
[0089]
Process 2: Correspondence to room with surrounding noise (floor noise is large) Process 2 is the
process 1 when the floor noise is large and the threshold level is updated automatically
automatically, the following measures are taken as a measure when it is difficult to detect speech
start and end .
The DSP 25 determines the threshold of the speech start detection level and the speech end
detection level based on the predicted floor noise level. The DSP 25 sets the speech start
threshold level to be larger than the speech end threshold level (for example, a difference of 3 dB
or more). The DSP 25 reads out the level value peak-held by the sound pressure level detector at
constant time intervals.
[0090]
According to this method, since this threshold setting has the same value for all the microphones,
it is possible to recognize the speech start with the same level of voice between the person with
the noise source on the back and the person without the noise source.
[0091]
Speech start determination processing 1, comparing the output level of the sound pressure level
detector corresponding to the six microphones with the threshold of the speech start level, and
determining that the speech start level is exceeded when the threshold of the speech start level is
exceeded.
When the output levels of the sound pressure level detectors corresponding to all the
microphones exceed the threshold of the speech start level, the DSP 25 determines that the
signal is from the reception and reproduction speaker 16 and does not determine that the speech
is started. This is because the distance between the reception and reproduction speaker 16 and
all the microphones MC1 to MC6 is the same, so the sound from the reception and reproduction
speaker 16 reaches all the microphones MC1 to MC6 almost equally.
[0092]
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30
Process 2, two uni-directional microphones (the microphone MC1 and the microphone MC1
having the directivity axes shifted by 180 degrees in the opposite direction) at 60 degrees
equiangular, radial and equally spaced arrangement for the six microphones illustrated in FIG.
Three pairs of MC4, microphones MC2 and MC5, and microphones MC3 and MC6) are used to
utilize the level difference between the two microphone signals. That is, the following operation
is performed.
[0093]
[Table 6] Absolute value of (signal level of microphone 1-signal level of microphone 4) ... [1]
Absolute value of (signal level of microphone 2-signal level of microphone 5) ... [2] Signal level of
3-signal level of the microphone 6) ... [3]
[0094]
The DSP 25 compares the absolute values [1], [2], and [3] with the threshold of the speech start
level, and determines that speech is started when the threshold of the speech start level is
exceeded.
In the case of this processing, since all absolute values do not become larger than the threshold
of the speech start level as in the processing 1 (since the sound from the reception reproduction
speaker 16 reaches all the microphones equally), the reception reproduction speaker 16 It is not
necessary to determine whether the sound is from the speaker or from the speaker.
[0095]
Speaker Direction Detection Process The characteristic of the unidirectional microphone
illustrated in FIG. 6 is used to detect the speaker direction. In the unidirectional microphone, the
frequency characteristic and the level characteristic change as illustrated in FIG. 6 according to
the arrival angle of the voice from the speaker to the microphone. The results are illustrated in
FIGS. 7A to 7C. 7 (A) to 7 (C) show a speaker placed at a predetermined distance from the
communication device 1, for example, 1.5 meters, and fast Fourier transform (FFT) of voice
collected by each microphone at constant time intervals Show the results. The X axis represents
frequency, the Y axis represents signal level, and the Z axis represents time. The horizontal line
represents the cutoff frequency of the band pass filter, and the level of the frequency band
10-04-2019
31
sandwiched by this line is the band pass of the five bands from the microphone signal level
conversion process described with reference to FIGS. -Data converted to sound pressure level
through a filter.
[0096]
A determination method applied as an actual process for detecting the direction of the speaker in
the speech apparatus 1 as one embodiment of the present invention will be described.
Appropriate weighting processing is performed on the output level of each bandpass filter (0 at 0
dBFs for 1 dB full span (1 dBFs) steps, 3 for -3 dBFs, or vice versa). The weighting step
determines the resolution of the process. The above weighting process is performed every one
sample clock, the weighted score of each microphone is added, and the average value is
calculated with a certain number of samples, and the small (large) microphone signal of the total
point is determined as the microphone facing the speaker Do. An image of this result is shown in
Table 7 below.
[0097]
[0098]
In this example illustrated in Table 7, the DSP 25 determines that the sound source is present
(speaker is present) in the direction of the first microphone MC1, since the smallest total point is
the first microphone MC1.
The DSP 25 holds the result in the form of a sound source direction microphone number. As
described above, the DSP 25 performs weighting on the output level of the band pass filter of the
frequency band for each microphone, and ranks the output signals of each band pass filter in the
order of small (or large) microphone signals. And determine that the microphone signal having
the first rank in three or more bands is the microphone facing the speaker. Then, assuming that
the sound source is present in the direction of the first microphone MC1 (speaker is present), the
DSP 25 creates a score sheet as shown in Table 8 below.
[0099]
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[0100]
Actually, the performance of the first microphone MC1 is not necessarily the best at all band pass
filter outputs due to sound reflections and standing waves due to the characteristics of the room,
but a majority of the five bands are If it is the first place, it can be determined that the sound
source is present (the speaker is present) in the direction of the first microphone MC1.
The DSP 25 holds the result in the form of a sound source direction microphone number.
[0101]
The DSP 25 sums up the output level data of each band pass filter of each microphone in the
form shown in Table 9 below, determines the microphone signal with a large level as the
microphone facing the speaker, and the result is the sound source direction microphone number
In the form of
[0102]
[Table 9] MIC1 Level = L1-1 + L1-2 + L1-3 + L1-4 + L1-5 MIC2 Level = L2-1 + L2-2 + L2-3 + L2-4
+ L2-5 MIC3 Level = L3-1 + L3-2 + L3-3 + L3-4 + L3-5 MIC4 Level = L4-1 + L4-2 + L4-3 + L4-4 +
L4-5 MIC5 Level = L5-1 + L5 -2 + L5-3 + L5-4 + L5-5 MIC6 Level = L6-1 + L6-2 + L6-3 + L6-4 +
L6-5
[0103]
Speaker Direction Microphone Switching Timing Determination Processing When activated by
the utterance start determination result in step 2 of FIG. 21 and when a new speaker microphone
is detected from the detection processing result of the speaker direction in step 3 and the past
selection information, The DSP 25 issues a command for switching the microphone signal to the
selection switching process of the microphone signal in step 5, and notifies the microphone
selection result display means 30 (light emitting diodes LED1 to 6) that the speaker microphone
has been switched, To notify that the communication device 1 has responded to his / her speech.
[0104]
In a room with a large echo, in order to remove the influence of reflected sound and standing
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waves, the DSP 25 prohibits the effect of a new microphone selection command unless the
speech end determination time (for example, 0.5 seconds) has elapsed after switching the
microphone.
From the microphone signal level conversion processing result of step 1 of FIG. 21 and the
detection processing result of the speaker direction of step 3, two microphone selection
switching timings are prepared in this embodiment.
[0105]
First method: When the start of speech can be clearly determined If the speech from the selected
microphone direction has ended and there is a new speech from another direction.
In this case, the DSP 25 starts speech after the speech termination judgment time (for example,
0.5 seconds) has elapsed after all the microphone signal levels (1) and the microphone signal
levels (2) have fallen below the speech termination threshold level. It is judged that the speech is
started when any microphone signal level (1) becomes equal to or higher than the speech start
threshold level, and the microphones facing the speaker direction are properly collected based
on the information of the sound source direction microphone number The microphone is
determined, and the microphone signal selection switching process of step 5 is started.
[0106]
2nd method: When a voice with a louder voice is newly input from another direction while the
message is being continued In this case, the DSP 25 ends the speech from the speech start (when
the microphone signal level (1) becomes equal to or higher than the threshold level) After the
judgment time (for example, 0.5 seconds) or more has elapsed, the judgment process is started.
If it is determined that the sound source direction microphone number from the processing in 3
is changed and stable before the speech end detection, the DSP 25 is more than the speaker
currently selected as the microphone corresponding to the sound source direction microphone
number. It is determined that there is a loud speaker and the sound source direction microphone
is determined to be a valid sound collection microphone, and the microphone signal selection
switching process of step 5 is activated.
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[0107]
Selection Switching Process of Microphone Signal Facing Detected Speaker The DSP 25 is
activated by a command selected and determined by the command from the switching timing
determination process of the microphone of the speaker direction in step 4 of FIG. The selection
switching process of the microphone signal of the DSP 25 is configured by six multipliers and a
six-input adder as illustrated in FIG. In order to select a microphone signal, the DSP 25 sets the
channel gain (channel gain: CH Gain) of the multiplier to which the microphone signal to be
selected is connected to [1] and the CH gain of the other multipliers to [0]. By doing so, the
processing result of (microphone signal О [1]) and (microphone signal О [0]) selected is added
to the adder, and a desired microphone selection signal is obtained at the output.
[0108]
As described above, when the channel gain is switched to [1] or [0], a click sound may be
generated due to the level difference of the microphone signal switched. Therefore, in the twoway communication device 1, as illustrated in FIG. 23, to change the change in CH Gain from [1]
to [0] and from [0] to [1], a switching transition time, for example, It is made to cross
continuously by changing it in a time of 10 ms to avoid generation of click sound due to the level
difference of the microphone signal.
[0109]
Further, by setting the maximum of the channel gain to other than [1], for example, [0.5], it is
possible to adjust the echo cancellation processing operation in the DSP 25 in the latter stage.
[0110]
As described above, the communication device according to the first embodiment of the present
invention is not affected by noise, and can be effectively applied to two-way conferences such as
a conference.
Of course, the communication device of the present invention is not limited to the conference,
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35
and can be applied to various other applications. That is, the communication device of the first
embodiment of the present invention is also suitable for measuring the voltage level in the pass
band when it is not necessary to emphasize the group delay characteristics of each pass band.
Therefore, for example, a simple spectrum analyzer, a level meter (FFT-like) that performs fast
Fourier transform (FFT) processing, a level detection processing apparatus for checking equalizer
processing results such as graphic equalizer, a car stereo, a level meter such as radio cassette
player, etc. It can be applied to
[0111]
The communication device of the first embodiment of the present invention has the following
advantages in terms of structure. (1) The positional relationship between a plurality of
unidirectional microphones and the reception and reproduction speaker is constant, and the
distance from the reception reproduction speaker is very short, so that the sound from the
reception and reproduction speaker passes through the conference room (room) environment.
The level coming back directly from the level coming back to multiple microphones is
overwhelmingly dominant. Therefore, the characteristics (signal level (intensity), frequency
characteristics (f characteristic), and phase) in which the sound reaches the microphones from
the reception and reproduction speaker are always the same. That is, in the speech apparatus of
the present invention, there is an advantage that the transfer function is always the same.
[0112]
(2) Therefore, there is no change in the transfer function when switching the microphone, and
there is an advantage that it is not necessary to adjust the gain of the microphone system each
time the microphone is switched. In other words, there is an advantage that it is not necessary to
redo once adjustment at the time of manufacturing the communication device.
[0113]
(3) Even if the microphones are switched for the same reason as described above, only one echo
canceller may be configured with a digital signal processor (DSP). The DSP is expensive, and the
space for arranging the DSP on a printed circuit board with various members mounted and a
small space may be small.
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[0114]
(4) Since the transfer function between the receiving and reproduction speaker and the plurality
of microphones is constant, there is an advantage that the sensitivity difference adjustment of the
microphone itself, which is ▒ 3 dB, can be performed by the unit alone.
[0115]
(4) The table on which the talking device is mounted usually uses a round table, but it is used as
a speaker system that evenly distributes (separates) voice of equal quality in all directions with
one receiving and reproducing speaker in the talking device Became possible.
[0116]
(5) The sound from the receiver / reproduction speaker is transmitted to the table surface
(boundary effect) to effectively and evenly reach the conference participants effectively and
evenly, and the sound from the opposite side in the ceiling direction of the conference room
There is an advantage that the phase cancellation is performed to make a small sound, and the
reflected sound from the ceiling direction to the conference participants is small, and as a result,
a clear sound is distributed to the participants.
[0117]
(6) Since the sound emitted from the receiving and reproducing speaker simultaneously reaches
all the plural microphones at the same volume, it is easy to determine whether it is the voice of
the speaker or the receiving voice.
As a result, erroneous determination of the microphone selection process is reduced.
[0118]
(7) Level comparison for direction detection can be easily performed by arranging an even
number of microphones radially at equal angles and at equal intervals.
[0119]
(8) To collect the microphone by vibration due to the sound of the receiving reproduction
speaker that can be transmitted through the printed circuit board on which the microphone is
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mounted by the damper using the buffer material, the flexible or resilient microphone supporting
member, or the like The impact can be reduced.
[0120]
(9) The sound of the reception and reproduction speaker does not directly enter the microphone.
Therefore, in this communication device, the influence of noise from the reception and
reproduction speaker is small.
[0121]
The speech apparatus according to the first embodiment of the present invention has the
following advantages in terms of signal processing.
(A) A plurality of unidirectional microphones are arranged radially at equal intervals to enable
detection of the sound source direction, and microphone signals are switched to collect (collect)
sound with good S / N and clear sound. , Can be sent to the other party.
(B) Sounds from surrounding speakers can be S / N well collected to automatically select a
microphone facing the speaker.
(C) In the present invention, signal analysis is simplified by dividing the passing audio frequency
band as a method of microphone selection processing and comparing the levels of the divided
frequency bands. (D) The microphone signal switching process of the present invention is
realized as signal processing of a DSP, and a click sound is not generated at the time of switching
by cross-fading a plurality of signals. (E) The microphone selection result can be notified to the
microphone selection result display means such as a light emitting diode, or to the outside.
Therefore, for example, it can also be used as speaker position information to a television camera.
[0122]
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Second Embodiment A technique for automatically adjusting a sensitivity difference of a
microphone will be described as a second embodiment of the microphone / speaker integrated
configuration / communication device (communication device) according to the present
invention.
[0123]
As a method of adjusting the gain of the microphone amplifier, a method of adjusting the gain of
the microphone analog amplifier to absorb the sensitivity difference between the microphones is
generally assumed, but in such a method, reflection or absorption of sound is considered. There
is a tendency for the influence of the coordinator and so on.
That is, the level of adjustment tends to be different between when the adjuster is close to the
microphone during adjustment and when the adjuster is away from the microphone. In addition,
such a method requires troublesome work such as connection and disconnection between the
output signal of the microphone amplifier and the measuring device. In the second embodiment
of the present invention, in order to overcome the problems described above, the sensitivity
difference of the microphone is automatically adjusted by the method described below.
[0124]
Adjustment of the sensitivity difference of the microphone of the second embodiment of the
present invention is based on the following concept. ?? The call device 1 according to the
embodiment of the present invention, for example, as shown in FIG. Therefore, if the reference
signal is line-in, it can be input to the DSP 26 and the DSP 25 via the A / D converter 274, so that
the sensitivity difference of the microphone can be adjusted without providing a special
measuring device. ?? The error range of the sensitivity difference can be freely set by the
program of the DSP 25. ?? By performing automatic adjustment, a nonstandard microphone
can be identified and a connection failure can be detected. Similarly, a defect or the like of an
amplification unit that amplifies a microphone signal is also detected.
[0125]
As a precondition, in the second embodiment, as illustrated in FIG. 4, even number of
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microphones, for example, six microphones, are equally spaced at equal angles radially and
equidistantly from the reception / playback speaker 16 in the second embodiment. It is arranged.
The positional relationship between the microphones MC1 to MC6 and the reception and
reproduction speaker 16 is as illustrated in FIG. 3 as to whether the reception and reproduction
speaker 16 is disposed below the microphones MC1 to MC6 or as illustrated in FIG. The
reception / playback speaker 16 may be disposed above the MC1 to MC6.
[0126]
Apparatus Configuration An apparatus configuration according to the second embodiment is
basically illustrated in FIG. 5, and the details are the configuration illustrated in FIG. In FIG. 24,
variable gain type amplifiers 301 to 306 for performing gain adjustment are actually disposed
between the microphones MC1 to MC6 and the A / D converters 271 to 273 in FIG. Alternatively,
the A / D converters 271 to 274 in FIG. 5 may be A / D converters 271 to 274 with variable gain
type amplifiers 301 to 306. The DSP 25 performs the various processes described above, but the
first to sixth variable attenuators (ATTs) 2511 to 2516 and the first to sixth level detectors 2521
to 2526 serve as parts for adjusting the sensitivity difference of the amplifiers 301 to 306. , A
level determination / gain control unit 253, and a test signal generation unit 254. The DSP 26
has an echo cancellation transmission processing unit 261 and an echo cancellation reception
unit 262.
[0127]
The variable gain type amplifiers 301 to 306 are amplifiers capable of changing the gain, and the
gain adjustment is performed by the level determination / gain control unit 253. However, when
the variable gain type amplifiers 301 to 306 are built in the A / D converters 271 to 273, gain
adjustment can not be freely performed. That is, there are cases where gain adjustment can be
freely performed, and there are also restrictions on the control width of variable gain amplifiers
301 to 306, etc. In this embodiment, the situation of variable gain amplifiers 301 to 306 Process
according to
[0128]
The variable attenuators 2511 to 2516 are also attenuators capable of changing the amount of
attenuation, and the level determination / gain control unit 253 performs control of the amount
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of attenuation by outputting attenuation coefficients 0.0 to 1.0. Note that, since the variable
attenuators 2511 to 2516 are processed in the DSP 25, in reality, the level determination / gain
control unit 253 in the same DSP 25 controls (adjusts) the attenuation value of the portions of
the variable attenuators 2511 to 2516. become.
[0129]
Each of the level detection units 2521 to 2526 is composed of a band pass filter 252a, an
absolute value calculation unit 252b, and a peak level detection and holding unit 252c, and
basically the same as the configuration illustrated in FIG. It is. The operation of the circuit
configuration illustrated in FIG. 17 has been described above.
[0130]
When a test sound is emitted from the sound level meter or the reception / playback speaker 16
in a room of a certain size (conference room), it is arranged at an equal distance d from the
soundness meter or the reception / playback speaker 16 unless there is a reflector or sound
absorber. An approximately equal signal arrives at each of the microphones MC1 to MC6
provided. Test voices from the noise level meter or the reception / playback speaker 16 collected
by the microphones MC1 to MC6 are amplified by the variable gain amplifiers 301 to 306,
converted to digital signals by the A / D converters 271 to 273, and then stored in the DSP 25. It
attenuates in the variable attenuators 2511 to 2516. The band pass filter 252a in the level
detection units 2521 to 2526 passes frequency components in a predetermined band, the
absolute value calculation unit 252b performs the calculation shown in Table 6, and the peak
level detection and holding unit 252c detects the maximum value. Being held. The level
determination / gain control unit 253 adjusts the attenuation amount (attenuation coefficient) of
the variable attenuation units 2511 to 2516 to adjust the sensitivity difference of each of the
microphones MC1 to MC6.
[0131]
Design Value of Sensitivity Difference Adjustment Error In the second embodiment, for example,
a microphone of ▒ 3 dB is assumed as a nominal error of the microphone sensitivity. In the
second embodiment, for example, 0.5 dB or less is targeted as a design value of the sensitivity
difference adjustment error. In addition, since it changes with the environments where a two-way
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41
communication apparatus is installed, about 0.5-1.0 dB is also appropriate as an actual sensitivity
difference adjustment error, for example.
[0132]
The test signal generation unit 254 inputs to the line input terminal the pink noise of the
reference input level (a sound pressure that is sufficiently large with respect to ambient noise is
generated), for example, 20 dB pink noise. put out. Alternatively, as shown by a broken line in
FIG. 24, the test signal output from the test signal generation unit 254 can be re-input to the DSP
25 via the echo cancellation transmission processing unit 261.
[0133]
The method of adjusting the microphone sensitivity difference is classified into cases 1 to 5
below according to the circuit configuration conditions of the variable gain amplifiers 301 to 306
and the like, and in the present embodiment, the process is divided into cases.
[0134]
Case 1: Since the variable gain type amplifiers 301 to 306 are not built in the A / D converters
271 to 273 but provided as independent amplifiers 301 to 306, the gain of the amplifiers 301 to
306 can be determined as the level of the DSP 25. When the gain control unit 253 can not
perform digital control: In this case, the level determination / gain control unit 253 adjusts the
attenuation value of the variable attenuators 2511 to 2516.
That is, the variable gain type amplifiers 301 to 306 are designed to obtain the minimum
required line output level when using the microphone with the lowest sensitivity, and the level
judgment / gain control unit 253 is variable attenuation. Adjust the attenuation value of the units
2511 to 2516.
[0135]
The processing of the level determination / gain control unit 253 will be described below with
reference to FIG. Step S201: The attenuation value of the variable attenuators 2511 to 2516 is
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set to 0 dB (1). Furthermore, it waits until the level detection operation of the level detection unit
252 is stabilized. Step S202: The average level of each microphone signal level-converted by the
level detection units 2521 to 2526 is measured. Steps S203 to S207: The attenuation values of
the variable attenuators 2511 to 2516 are changed so that each channel becomes the design
value level of the sensitivity difference adjustment error with reference to the measured average
value. Also, using the average level of each microphone signal level-converted by the first to sixth
level detection units 2521 to 2526 after changing the attenuation value of the variable
attenuation units 2511 to 2516, each channel is repeatedly different in sensitivity The
attenuation values of the variable attenuators 2511 to 2516 are changed so as to reach the
design value level of the adjustment error. The accuracy of adjusting the sensitivity difference is
determined by the accuracy of tracking the level difference at this time. By thus determining in
advance the adjustment range of the attenuation value, it is possible to detect a defect in the
microphone.
[0136]
Case 2: The gains of the variable gain amplifiers 301 to 306 can be digitally controlled for each
channel, and the control width is a sensitivity difference adjustment error, for example, 0.5 dB or
less: As illustrated in FIG. 26, level determination The gain control unit 253 performs the
following process of adjusting the gains of the variable gain amplifiers 301 to 306.
[0137]
Step S211: The gains of the variable gain type amplifiers 301 to 306 are set to initial values.
Further, the attenuation values of the variable attenuators 2511 to 2516 are set to 0 dB (1), and
the level detection units 2521 to 2526 are on standby until level detection is stabilized. Step
S212: The average value of each microphone level-converted by the level detection units 2521 to
2526 is measured. Steps S213-219: If there is a microphone of a channel whose measurement
result falls within the ▒ 0.5 dB value which is the design value of sensitivity difference
adjustment error, the adjustment of that channel is ended. Otherwise, the gains of the variable
gain amplifiers 301 to 306 of each microphone are changed (adjusted) to fall within the design
value range of the sensitivity difference adjustment error. Also, using the average level of each of
the microphone signals level-converted by the level detection units 2521 to 2526 after changing
the gain of the variable gain type amplifiers 301 to 306, each channel is repeatedly designed for
a sensitivity difference adjustment error The gains of the variable gain amplifiers 301 to 306 are
changed to be at the level. As described above, by previously determining the adjustment range
of the gains of the variable gain amplifiers 301 to 306, it is possible to detect the variable gain
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amplifiers 301 to 306 or the microphone failure.
[0138]
Case 3: The gains of the variable gain amplifiers 301 to 306 can be digitally controlled for each
channel, and the control width is, for example, 2 dB or more: As illustrated in FIG. 27, the level
determination / gain control unit 253 First, the gain adjustment of the variable gain type
amplifiers 301 to 306 is performed (steps S231 to S237), and then the attenuation amount of
the variable attenuators 2511 to 2516 is adjusted (steps S238 to S241).
[0139]
Steps S231 to S238: Basically, the process is the same as in the case 2 described with reference
to FIG. 26, and the gains of the variable gain amplifiers 301 to 306 are adjusted.
That is, in step S231, the gains of the variable gain amplifiers 301 to 306 are set to initial values,
the attenuation value of the variable attenuators 2511 to 2516 is set to 0 dB (1), and level
conversion is performed by the level detection units 2521 to 2526. Measure the average value of
each microphone. If there is a microphone of a channel whose measurement result falls within ▒
0.5 dB, which is the designed value of sensitivity difference adjustment error, adjustment of that
channel is ended. Otherwise, set the gains of the variable gain amplifiers 301 to 306 and set the
gains of the remaining variable gain amplifiers 301 to 306 so that the average level falls within
the positive value of the sensitivity difference adjustment error design value. .
[0140]
In case 3, the control width of the gain adjustment of the variable gain type amplifiers 301 to
306 is 2 dB, and the control width as in case 2 is not 0.5 dB. Therefore, the amount of
attenuation is adjusted by the variable attenuation units 2511 to 2516 by the following
processing.
[0141]
Steps S240 to S243: Change the attenuation amount of the variable attenuators 2511 to 2516 of
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the microphone signals of the channels not included in the design value of the sensitivity
difference adjustment error, and wait until the levels in the level detection units 2521 to 2526
become stable. Level is taken in the level of my signal which is stabilized, average value
processing is done, repetition processing is done until the value is within the design value range
of sensitivity difference adjustment error, average level value of microphone signal channel
sensitivity difference The attenuation values of the variable attenuators 2511 to 2516 are set so
as to be within ▒ 0.5 dB of the design value of the adjustment error. As described above, by
previously determining the attenuation value and the adjustment range of the gains of the
variable gain amplifiers 301 to 306, it is possible to detect a defect in the variable gain amplifiers
301 to 306 or the microphone.
[0142]
Case 4: The variable gain type amplifiers 301 to 306 are incorporated in the A / D converters
271 to 273. The gains of the amplifiers 301 to 306 can actually be digitally controlled only at
two channels simultaneously, and the control width is adjusted for sensitivity difference Error,
for example, 0.5 dB or less: As illustrated in FIGS. 28 and 29, the level determination / gain
control unit 253 performs the following process.
[0143]
Steps S251 and S271: The gains of the variable gain amplifiers 301 to 306 are set to initial
values, the attenuation value of the variable attenuators 2511 to 2516 is set to 0 dB (1), and the
level detection of the level detectors 2521 to 2526 is stable. Wait until you do.
Steps S252 and S272: An average value process of the level detection detected by the level
detection units 2521 to 2526 is performed.
[0144]
Hereinafter, as illustrated in FIGS. 28 and 29, the following two adjustment methods are adopted.
FIG. 28 shows a method of performing gain adjustment of the variable gain type amplifiers 301
to 306 first and adjusting the attenuation of the variable attenuators 2511 to 2516 later (case 41), and FIG. 29 is illustrated in FIG. Contrary to the method described above, the adjustment of
the attenuation amount of the variable attenuators 2511 to 2516 is performed first, and the gain
adjustment of the variable gain amplifiers 301 to 306 is performed later (case 4-2).
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[0145]
Case 4-1: As illustrated in steps S253 to S259 of FIG. 28, the variable gain type amplifiers 301 to
306 can be set to the signal levels of channels with low signal levels in the group where the gain
can be set, and the signal levels of other channels. The gains of the variable gain amplifiers 301
to 306 are adjusted to fall within the low channel signal level ▒ 0.5 dB. Next, as illustrated in
steps S261 to S264, the attenuation values of the variable attenuators 2511 to 2516 are
adjusted so that the higher signal level falls within ▒ 0.5 dB of the design value of the sensitivity
difference adjustment error.
[0146]
Case 4-2: As illustrated in steps S273 to S277 in FIG. 29, the gains of the variable gain amplifiers
301 to 306 are adjusted so that the average level value of the microphone signal channel falls
within ▒ 0.5 dB of the design value. Next, as illustrated in steps S278 to S282, the signal level of
the channel in the group where the gain of the variable gain type amplifiers 301 to 306 can be
set is low, and the signal level of the other channel is low ▒ 0. Adjust the gains of the variable
gain amplifiers 301-306 to fall within 5 dB.
[0147]
As described above, by previously determining the attenuation values of the variable attenuators
2511 to 2516 and the adjustment ranges of the gains of the variable gain amplifiers 301 to 306,
it is possible to detect a defect in the variable gain amplifiers 301 to 306 or the microphone.
[0148]
Case 5: The variable gain type amplifiers 301 to 306 are incorporated in the A / D converters
271 to 273. The gains of the amplifiers 301 to 306 can actually be digitally controlled only at
two channels simultaneously, and the control width is, for example, In the case of 2 dB or less: As
illustrated in FIG. 30, the level determination / gain control unit 253 first adjusts the attenuation
amount of the variable attenuation units 2511 to 2516 (S293 to S297), and then the variable
gain amplifier 301 The gain adjustment of -306 is performed (S298-S303), and the attenuation
amount of the variable attenuators 2511 to 2516 is further adjusted (S304-S308).
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46
Details will be described below.
[0149]
Step S291: The gains of the variable gain amplifiers 301 to 306 are set to initial values, the
attenuation values of the variable attenuators 2511 to 2516 are set to 0 dB (1), and standby is
performed until level detection by the level detectors 2521 to 2526 is stabilized. Do. Step S292:
The average value process of each microphone signal level-converted by the level detection units
2521-2526 is performed. Steps S293 to S297: Adjust the attenuation value of the variable
attenuators 2511 to 2516 so that the signal level of the lowest level of the microphone channel
in the group in which the gains of the variable gain amplifiers 301 to 306 can be set match the
other signal level Do. Steps S298 to S303: The gains of the variable gain type amplifiers 301 to
306 are adjusted such that the average level value of the microphone signal channel falls within
▒ 1 dB of the design value of the sensitivity difference adjustment error. Steps S304 to S308:
The attenuation values of the variable attenuators 2511 to 2516 are adjusted again so that the
microphone signal level becomes ▒ 0.5 dB of the design value of the sensitivity difference
adjustment error. As described above, by previously determining the attenuation value and the
adjustment range of the gains of the variable gain amplifiers 301 to 306, it is possible to detect a
defect in the circuit or the microphone.
[0150]
According to the second embodiment, a plurality of microphones disposed at equal distances
from the reception / playback speaker 16 automatically adjust the sensitivity difference of the
pair of opposing microphones fixedly connected to the microphone amplifier. The gain difference
of the transmitting microphone can be automatically adjusted so that the acoustic coupling
between the receiving and reproducing speaker 16 and each of the sound collecting microphones
MC1 to MC6 becomes equal.
[0151]
In the implementation of the present embodiment, it is only necessary to use the microphone /
speaker integrated configuration / communication device itself without requiring a special
device.
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Therefore, the above adjustment can be performed in a state in which the microphone-speaker
integrated configuration type / communication device is provided.
[0152]
Third Embodiment A third embodiment of the present invention will be described with reference
to FIGS. According to the third embodiment of the present invention, in the two-way
communication system, the difference in sensitivity between a plurality of transmitting
microphones equidistant from the receiving speaker is automatically corrected, and the acoustic
coupling between the speaker and the collecting microphone becomes equal. A method and
apparatus for automatically adjusting the amplifier gain of a microphone digitally.
[0153]
In a device in which a plurality of microphones MC1 to MC6 are selected and switched and used
as in the case of the communication device 1 according to the embodiment of the present
invention, variations in signal level due to sensitivity differences of the microphones MC1 to MC6
become a problem. is there. In such a case, the output volume or gain of the amplifiers of the
analog microphones MC1 to MC6 is adjusted to make the output levels of the microphone signals
uniform, and the output levels of the analog line amplifiers are adjusted by the output level or
gain adjustment. Is common.
[0154]
As such a method, there is known a method of adjusting a gain of an analog microphone
amplifier to absorb a microphone sensitivity difference. Although this method is the most
common method, it has the disadvantage of being easily influenced by the coordinator, such as
reflection and absorption of sound. For example, the level is likely to differ between when the
coordinator is close and away during the adjustment. In addition, complicated operations such as
switching between the microphone amplifier output and the measuring instrument are required.
The third embodiment of the present invention improves such a problem.
[0155]
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48
Concept of Third Embodiment FIG. 31 is a block diagram of a communication device according to
a third embodiment of the present invention. As illustrated in FIG. 3 and FIG. 31, the
communication device of the embodiment of the present invention is provided with the speaker
16. Therefore, if the sound source test signal from the second test signal generation unit 258
described later is input to the terminal terminal LINE-IN of the telephone set 1 and signal
processing is performed in the DSP 25 having various signal processing functions, the speaker
16 and each component are processed. The degree of acoustic coupling with the microphones
MC1 to MC6 can be specified, and the degree of acoustic coupling can be adjusted. In particular,
the DSP 25 can also be used as a test signal generator. As a result, the degree of acoustic
coupling between the speaker 16 and the microphones MC1 to MC6 is automatically measured
without using a special measuring instrument such as a test signal generator, a sound measuring
instrument, an analyzer, etc. By adjusting the components 301 to 306, the variable attenuators
2511 to 2516, and the like, the degree of acoustic coupling can be adjusted. Further, the total
gain of the transmission system can be automatically adjusted by the circuit configuration.
Furthermore, the adjustment error can be freely set by the program.
[0156]
Device Configuration As illustrated in FIG. 4, six unidirectional microphones are radially disposed
at an interval of 60 degrees, and the speaker 16 is positioned above or below the microphone as
illustrated in FIG. 3 or 8. The distance between each of the microphones MC1 to MC6 and the
speaker 16 is assumed to be equal.
[0157]
DSP Although the communication device 1 illustrated in FIG. 31 is similar to the communication
device 1 illustrated in FIG. 24 in the second embodiment, the configuration and processing
content in the first digital signal processor (DSP) DSP 25 are different. .
The DSP 25 in FIG. 31 is a second level determination / gain control unit 257, and a second test
signal generation unit instead of the level determination / gain control unit 253 and the test
signal generation unit 254 in the DSP 25 illustrated in FIG. And H.258. That is, the second level
determination / gain control unit 257 performs the process described with reference to FIGS. 33
to 40, and the second test signal generation unit 258 generates a test signal different from that
of the test signal generation unit 254. The DSP 25 is provided with switch means for outputting
the signal of the selected microphones MC1 to MC6 to LINE OUT as in the above-described
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49
embodiment. In other words, the second level determination / gain control unit 257 has a
function of selecting the microphone signal by the method described above. Furthermore, the
second level determination / gain control unit 257 of the DSP 25 has amplification means for
digitally amplifying the signal selected by the switch means, and in the second level
determination / gain control unit 257, D / A The gain of the signal output to LINEOUT via the
converter 281 can be adjusted. As such a gain adjustment method, the second level
determination / gain control unit 257 performs multiplication. The level of the signal subjected
to gain adjustment and output to LINE OUT via the D / A converter 281 is detected by a level
detection unit 391 described later. The other configuration of the DSP 25, for example, the digital
variable attenuation unit 251, the digital level detection unit 252, and the like are the same as
those described with reference to FIG.
[0158]
Level Detection Unit A level detection unit 391 for detecting the output of the output amplifier
291 is added. The level detection unit 391 is a peak level detector for signal level measurement,
and in the present embodiment, serves as a measurement instrument for automatic adjustment.
The level detection unit 391 may have, for example, a circuit configuration similar to the circuit
configuration of the level detection unit 252 in the DSP 25, for example, an absolute value
calculation unit and a peak level detector. However, although the level detection unit 252 in the
DSP 25 is realized by digital signal processing, the level detection unit 391 is configured by a
hardware circuit, but the function is the same as that of the level detection unit 252. In the level
detection unit 391, a band pass filter corresponding to the band pass filter unit 252a may be
added to the front stage of the absolute value calculation unit 252b of the level detection unit
252.
[0159]
A / D Converters Each of the A / D converters 271 to 274 has a function of receiving a twomicrophone signal and converting it into a digital signal, as in the above-described example. Also,
from the viewpoint of digitally adjusting the gain of the input microphone signal, A / D
converters 271 to 274 with digital variable gain are used.
[0160]
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50
Microphone Signal Selection Method (Combination) Two microphone signals can be input to each
of the A / D converters 271 to 274. Further, since the microphones MC1 to MC6 are equidistant
to the speaker 16 and under the same condition, two microphone signals to be input to the A / D
converters 271 to 274 can be arbitrarily selected. From the viewpoint of the first and second
embodiments, it is necessary to input microphone pairs which are opposed and arranged in a
straight line sandwiching the center C of the printed circuit board 21 into the same A / D
converter. Also in this embodiment, such an input method may be used, and in the case where
other applications are considered, wiring change is also unnecessary, which is convenient.
However, in consideration of the present embodiment, any combination can be input to each A /
D converter without such limitation. As an example, in the arrangement of the microphones MC1
to MC6 illustrated in FIG. 4, signals of two adjacent microphones can also be input to each A / D
converter. Such an example is described below.
[0161]
[Table 10] MIC1, MIC2-ADC1 MIC3, MIC4-ADC2 MIC5, MIC6-ADC3
[0162]
In addition, when the 1 input 1 output type A / D converter is used, the combination of the signal
mentioned above is out of object of discussion.
[0163]
In FIG. 31, microphones MC1 to MC6 to A / D converter block 27 are used as a transmission
signal input unit B1, D / A converter 281 to amplifier 291 as a transmission output unit B2, and
D / A converter 282 to a speaker. The reception output unit B3 is used as the reception output
unit B3, and the reception input unit B4 is used as the LINEIN to A / D converter 274.
[0164]
The variable attenuation unit 251 is configured such that the amount of attenuation can be
changed by the second level determination / gain control unit 257.
The variable gain type amplifiers 301 to 306 can also adjust the gain by the second level
determination / gain control unit 257, but in the present embodiment, the case of automatically
adjusting the gain digitally will be described. The digitally adjustable gain portions in D
converters 271-274 are adjusted.
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51
Therefore, variable gain type amplifiers 301 to 306 are irrelevant in the present embodiment.
[0165]
FIG. 32 shows the level diagram of the signal required.
Since the receiver gain setting requires a measuring instrument, it is not the target of automatic
adjustment, and is left as the design value. The gain of the design value is set so that the sound
pressure level becomes 70 dB at a point 2 m away from the receiving speaker 16 when pink
noise of 20 dB is input to the receiving input terminal. It is particularly important in the gain
setting of the transmission system that the microphone output level of the echo cancellation
processing block is -25 dB from the speaker 16 through the microphone under the above
conditions with respect to the reception power level of -20 dB of the echo cancellation
processing block in the DSP 26 It is to determine the signal level of the transmission system
including the microphone selection process in the DSP 25.
[0166]
Second Test Signal Generation Unit The second test signal generation unit 258 may output a pink
noise signal, a white noise signal, or the like, as in the test signal generation unit 254, for
example. It can also output a signal. The sound source test signal generated by the second test
signal generation unit 258 is input to the input terminal LINE IN of the communication device 1.
[0167]
Processing of Second Level Judgment / Gain Control Unit The processing performed by the
second level judgment / gain control unit 257 can be roughly classified into the following three
processes as illustrated in FIG.
[0168]
??
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52
S411 to S413 1.1 S411: Microphone pair sensitivity difference adjustment 1 The sensitivity
difference between the microphones connected to the ADC 1 is adjusted. The magnitudes are
compared using masked values of lower 8 bits of 16 bit data obtained from the level detector.
The attenuation value (ATT value) of the variable attenuation unit 251 is increased in units of
one step 0x100 (HEX). Details of the process are shown in the flowchart of FIG.
[0169]
1.2 S412: Microphone pair sensitivity difference adjustment 2 Adjust the sensitivity difference
between the microphones connected to ADC2. The magnitudes are compared with masked values
of lower 8 bits of 16 bit data obtained from the level detector. The ATT value is increased by 1
step 0x100 (HEX). Details of the process are shown in the flowchart of FIG.
[0170]
1.3 S413: Microphone pair sensitivity difference adjustment 3 Adjust the sensitivity difference
between the microphones connected to ADC3. The magnitudes are compared with masked values
of lower 8 bits of 16 bit data obtained from the level detector. The ATT value is increased by 1
step 0x100 (HEX). Details of the process are shown in the flowchart of FIG.
[0171]
The processing illustrated in FIG. 34 to FIG. 36 is the attenuation amount of the variable
attenuators 2511 to 2516 for the higher level signal of the two signals input to and converted by
the respective A / D converters 271 to 274. And lower the level to bring both into a
predetermined range. The reason is that since the A / D converters 271 to 274 adjust the
common gain with two inputs, they can not be adjusted individually by the A / D converters 271
to 274. This is to adjust the level of the two inputs to a predetermined range. If the A / D
converters 271 to 274 are 1-input 1-conversion A / D converters 271 to 274, such processing is
not necessary.
[0172]
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53
?? Gain adjustment of microphone amplifier, S421 to S412 This is a part for correcting the
attenuation amount in the variable attenuation unit 251 in units of two microphone pairs, and
adjusts the gain of the microphone amplifier in the A / D converters 271 to 274. The microphone
amplifier gain adjustment realizes that the other microphone amplifier gains are increased to
match the microphone pair output level of the maximum value of the microphone pair output.
Details of the process are shown in the flowchart of FIG.
[0173]
?? Microphone Selection Unit Output Level Adjustment: S431 to S438 The signal output to the
external output terminal LINE OUT at which one microphone signal is selected is adjusted. The
second level determination / gain control unit 257 receives a signal whose level is detected by
the D / A converter 281 by the level detection unit 391, and adjusts the gain of the output
amplifier 291 to be equivalent to -25 dB, In order to adjust the gain of D / A converter 281 or to
adjust the gain of the signal after microphone signal selection in DSP 25, a predetermined
coefficient is added to the signal outputted to selected D / A converter 281. Ride. A method of
multiplying the signal output to the selected D / A converter 281 by a predetermined coefficient
can be easily realized digitally to adjust the gain of the signal after microphone signal selection in
the DSP 25.
[0174]
Adjustable Parts The parts that can be adjusted according to the third embodiment are shown
below. ?? Transmission System 1.1 Digital Gain of A / D Converters 271-274 The gain changes
simultaneously for two inputs. 1.2 Amount of attenuation of digital variable attenuation section
251 1.3 Digital gain of microphone selection signal performed by second level determination /
gain control section 257 1.4 Gain of D / A converter 281 Reception system 2.1 Gain of A / D
converter 274 connected to echo cancellation receiver 262
[0175]
That is, the embodiment of the present invention is not limited to the above-described
embodiment, and the following adjustment can be appropriately selected and performed. For
10-04-2019
54
example, only the variable attenuators 2511 to 2516 may be adjusted. In addition, if the A / D
converters 271 to 274 are 1-input 1-conversion type A / D converters, the levels of the two
signals in S411 (FIGS. 34 to 36) are not adjusted in level. The A / D converter or variable
attenuators 2511 to 2516 can be individually adjusted without performing gain adjustment
every two. Whether the gain adjustment of the A / D converter is performed first or the
attenuation adjustment of the variable attenuators 2511 to 2516 is performed first is determined
by determining a predetermined rule. For example, if the level is higher than a certain level, the
attenuation amount of the variable attenuators 2511 to 2516 is first adjusted, then the gain of
the A / D converter is adjusted, and in the opposite case, the gain is increased with the A / D
converter. After that, the attenuation amount of the variable attenuators 2511 to 2516 is
adjusted.
[0176]
As described above, according to the third embodiment, the digitally adjustable portion of the
telephone set 1 can be automatically adjusted without using any special measuring device or test
signal generating device.
[0177]
Modification of Third Embodiment FIG. 41 shows a modification of the third embodiment
illustrated in FIG.
The difference between the configuration of FIG. 41 and the configuration of FIG. 31 is that, in
FIG. 31, there is one input terminal of the echo cancellation receiver 262, while in FIG. The
number of connections to the input terminal of the portion 262 is increased to two. With the
circuit configuration of FIG. 41, the output signal from the echo cancellation transmission
processing unit 261 is input to the echo cancellation reception unit 262, echo cancellation of the
transmission signal is completely performed, and the influence can be eliminated. More accurate
sensitivity adjustment is possible.
[0178]
FIG. 1 (A) is a view showing an outline of a conference system as an example to which a
microphone / speaker integrated configuration / communication device (communication device)
of the present invention is applied, and FIG. 1 (B) is a view of FIG. FIG. 1 (C) is a diagram showing
10-04-2019
55
the arrangement of the telephone set and the conference participants placed on the table. FIG. 2
is a perspective view of the communication device according to the embodiment of the present
invention. FIG. 3 is an internal sectional view of the communication device illustrated in FIG. FIG.
4 is a plan view of the microphone / electronic circuit housing with the top cover of the
communication device illustrated in FIG. 1 removed. FIG. 5 is a diagram showing the connection
state of the main circuits of the microphone / electronic circuit housing unit, showing the
connection state of the connection of the first DSP and the second DSP. FIG. 6 is a characteristic
diagram of the microphone illustrated in FIG. FIGS. 7A to 7D are graphs showing the results of
analyzing the directivity of the microphone having the characteristics illustrated in FIG. FIG. 8 is a
partial configuration diagram of a modification of the speech apparatus of the present invention.
FIG. 9 is a graph showing an outline of the entire processing content in the first DSP. FIG. 10 is a
flowchart showing a first embodiment of the noise measurement method in the present
invention. FIG. 11 is a flow chart showing a second embodiment of the noise measurement
method in the present invention. FIG. 12 is a flow chart showing a third mode of the noise
measurement method in the present invention. FIG. 13 is a flowchart showing a fourth
embodiment of the noise measurement method in the present invention. FIG. 14 is a flow chart
showing a fifth mode of the noise measurement method in the present invention. FIG. 15 is a
view showing a filtering process in the speech apparatus of the present invention. FIG. 16 is a
frequency characteristic diagram showing the processing result of FIG. FIG. 17 is a block diagram
showing band pass filtering processing and level conversion processing of the present invention.
FIG. 18 is a flowchart showing the process of FIG. FIG. 19 is a graph showing a process of
determining the speech start and end in the speech apparatus of the present invention. FIG. 20 is
a graph showing a flow of normal processing in the speech apparatus of the present invention.
FIG. 21 is a flow chart showing the flow of the normal processing in the speech apparatus of the
present invention. FIG. 22 is a block diagram illustrating microphone switching processing in the
speech apparatus of the present invention. FIG. 23 is a block diagram illustrating a method of
microphone switching processing in the speech apparatus of the present invention. FIG. 24 is a
block diagram illustrating a partial configuration of the communication device of the second
embodiment of the present invention.
FIG. 25 is a flowchart showing a first processing method of the second embodiment of the
present invention. FIG. 26 is a flow chart showing a second processing method of the second
embodiment of the present invention. FIG. 27 is a flowchart showing a third processing method
of the second embodiment of the present invention. FIG. 28 is a flow chart showing a fourth
processing method of the second embodiment of the present invention. FIG. 29 is a flowchart
showing a second two processing methods of the second embodiment of the present invention.
FIG. 30 is a flowchart showing the fifth processing method of the second embodiment of the
present invention. FIG. 31 is a block diagram illustrating a partial configuration of the
communication device according to the third embodiment of the present invention. FIG. 32 is a
diagram illustrating the conditions in the third embodiment of the present invention. FIG. 33 is a
10-04-2019
56
flowchart showing the processing method of the third embodiment of the present invention. FIG.
34 is a flow chart showing a first detail of a part of FIG. FIG. 35 is a flow chart showing a second
detail of a portion of FIG. FIG. 36 is a flowchart showing third details of a portion of FIG. FIG. 37
is a flowchart showing fourth details of a portion of FIG. FIG. 38 is a flow chart showing a first
detail of a portion of FIG. FIG. 39 is a flow chart showing a second detail of a portion of FIG. FIG.
40 is a flow chart showing a third detail of a portion of FIG. FIG. 41 is a block diagram illustrating
a partial configuration of a modification of the communication device of the third embodiment of
the present invention.
Explanation of sign
[0179]
1 и и Microphone и speaker integrated configuration и communication device (communication
device) 11 и и и upper cover 12 и и и sound reflection plate 12a и и и sound reflection surface 12b и и
restraint member fixing portion 13 и и connection member 14 и и speaker Housing portion 14a иии
Sound reflecting surface, 14b и и Bottom surface 14c и и Upper surface 14b и 14d и и Internal cavity
14e и и Restraint member lower fixing portion 14f и и Restraint member penetration portion 15 и и
Operating portion 16 и и и и и и и 17 и и restraining member 18 и и damper 2 и и microphone и electronic
circuit housing portion 21 и и и printed circuit board MC1-MC и и microphone 22 и и microphone
supporting member 22a и и first microphone supporting member 22b и и second Microphone
support member 23 иии Microprocessor, 24 и Codec 25 и и First digital signal processor (DSP1) 301
... Variable gain type amplifier 251... Variable attenuation unit 252 .. Level detection unit 253 ..
Level determination / gain control unit 254 .. Test signal generation unit 257 .. Second level
determination / gain control unit 258. Second test signal generation unit 391 .. Level detection
unit 26... Second digital signal processor (DSP 2) 261 .. Echo cancellation transmission
processing unit 262 .. Echo cancellation receiver 27 .. A / D converter block 28 и и D / A converter
block 29 и и amplifier block 30 и и Microphone selection result display means LED 1 to 6 и и light
emitting diode
10-04-2019
57
l processing of the
communication device 1.
[0081]
As illustrated in FIG. 20, the communication device 1 performs audio signal monitoring
processing according to the sound collection signals from the microphones MC1 to MC6,
performs speech start / end determination, performs speech direction determination, and
performs microphone selection. The result is displayed on the microphone selection result
display means 30, for example, the light emitting diodes LED1 to LED6. Hereinafter, the
operation will be described mainly with the DSP 25 in the communication device 1 with
reference to the flowchart in FIG. The overall control of the microphone / electronic circuit
housing unit 2 is performed by the microprocessor 23, but the processing of the DSP 25 will be
mainly described.
10-04-2019
27
[0082]
Step 1: Monitoring of Level Conversion Signal The signals collected by the microphones MC1 to
MC6 are respectively transmitted to the band pass filter block 201 and the level conversion block
202 described with reference to FIGS. Since the seven types of level data are converted, the DSP
25 constantly monitors seven types of signals for each microphone sound collection signal.
Based on the monitoring result, the DSP 25 shifts to any one of the speaker direction detection
process 1, the speaker direction detection process 2, and the speech start / end determination
process.
[0083]
Step 2: Speech start / end judgment processing The DSP 25 judges the start / end of the speech
according to the method described in detail below with reference to FIG. When the DSP 25
detects the speech start, it notifies the judgment process of the direction of the speaker of step 4
that the speech start is detected. In addition, when the speech level is smaller than the speech
end level, the process of judging the start and end of the speech in step 2 starts a timer for the
speech end judgment time (for example, 0.5 seconds) and the speech end judgment time and the
speech level are speech When it is lower than the end level, it is determined that the speech is
ended. If it becomes higher than the speech end level within the speech end determination time,
the process of waiting is made until it becomes smaller than the speech end level.
[0084]
Step 3: Speaker Direction Detection Process The speaker direction detection process in the DSP
25 is performed by continuously searching the speaker direction. Thereafter, the data is supplied
to the determination process of the speaker direction in step 4.
[0085]
Step 4: Speaker Direction Microphone Switching Processing The timing determination processing
in the processing for switching the speaker direction microphone to the DSP 25 has been
selected from the speaker detection direction at that time based on the processing in Step 2 and
the processing in Step 3. When the speaker direction is different, the microphone selection of the
10-04-2019
28
new speaker direction is instructed to the microphone signal switching process of step 4.
However, if the chairman's microphone is set from the operation unit 15 and the chairman's
microphone and other conference participants speak simultaneously, priority is given to the
chairman's speech. At this time, the selected microphone information is displayed on the
microphone selection result display means 30, for example, the light emitting diodes LED1 to
LED6.
[0086]
Step 5: Transmission of Microphone Sound Collection Signal In the microphone signal switching
processing, only the microphone signal selected by the processing in step 4 out of the six
microphone signals is used as a transmission signal, and the other party from the telephone
apparatus 1 through the telephone line 920 To the line unit of the telephone line 920 illustrated
in FIG.
[0087]
Setting of speech start level threshold and speech end threshold Process 1: Measure floor noise
for a predetermined time, for example, one second, of each microphone immediately after power
on.
The DSP 25 reads out the peak-held level value of the sound pressure level detection unit at
constant time intervals, for example, 10 mSec intervals in this embodiment, and calculates an
average value of values for a predetermined time, for example, 1 minute, and floor noise I
assume. The DSP 25 determines the speech start detection level (floor noise +9 dB) and the
speech end detection level threshold (floor noise +6 dB) based on the measured floor noise level.
The DSP 25 subsequently reads out the peak-held level value of the sound pressure level detector
at constant time intervals. When it is determined that the speech is ended, the DSP 25 works as a
measurement of floor noise, detects the start of speech, and updates the threshold of the
detection level of the speech end.
[0088]
According to this method, since the threshold setting is different for each floor noise level at the
position where the mi
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