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JP2010081124

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DESCRIPTION JP2010081124
PROBLEM TO BE SOLVED: To provide a calibration method for an interphone device capable of
simply performing calibration of a microphone of an intercom device which aims to prevent
howling by canceling a speaker sound from a transmission signal using a plurality of
microphones. SOLUTION: Amplitude ratio data derivation for deriving data of an amplitude ratio
of each audio signal of the microphones M1 and M2 with respect to sound from the speaker SP
and driving steps S2 to S7 for driving the speaker SP of the intercom device A at a specific
frequency Step S8, a delay time data deriving step (S9) for deriving delay time data based on the
phase of each audio signal of the microphones M1 and M2 with respect to the voice from the
speaker SP; each data of the delay time and the amplitude ratio And an optimum coefficient
calculation step (S10) of calculating an optimum coefficient for the calculation process for
canceling the speaker sound, and a storage step (S11) of storing the calculated optimum
coefficient in the storage means. [Selected figure] Figure 1
Calibration method for intercom device
[0001]
The present invention relates to a calibration method for an intercom apparatus.
[0002]
2. Description of the Related Art Conventionally, there is an intercom apparatus that performs
two-way communication, and is provided with a speaker that outputs sound from an intercom
apparatus installed at another place, a microphone that inputs sound transmitted to the other
intercom apparatus, and the like.
04-05-2019
1
[0003]
And since howling occurs when the sound generated from the speaker gets into the microphone,
various measures against howling are taken.
For example, a delay circuit having a speaker and a pair of microphones and delaying the output
of the microphone closer to the speaker by the delay time of the sound wave corresponding to
the difference between the distance between the two microphones and the speaker; A level
adjustment amplifier circuit for matching the output levels of the microphones and a differential
amplifier circuit for which both outputs of both microphones passed through the delay circuit
and the level adjustment amplifier circuit are input, and the output of the differential amplifier
circuit is transmitted There has been proposed an intercom apparatus which cancels a speaker
sound from a transmission signal to prevent howling by setting the signal as a signal.
[0004]
In this intercom system, after both microphones pick up the voice from the speaker, delay and
level adjustment are performed, and the voice component from the speaker input to both
microphones is canceled by the differential amplifier circuit, thereby the voice from the speaker
Only the components are removed (cancellation processing) to try to prevent howling (see, for
example, Patent Document 1).
[0005]
In such an intercom apparatus having a plurality of microphones, it is necessary to calibrate the
sensitivity etc. of each microphone to ensure the cancellation effect of the speaker sound, and for
example, calibration is periodically performed in the field actually used. A method (see, for
example, Patent Document 2) and a method of performing calibration by switching a plurality of
ranges in a manufacturing process have been proposed (see, for example, Patent Document 3).
Patent No. 3226121 Japanese Patent Laid-Open No. 2004-343700 Japanese Patent Laid-Open
No. 2006-135551
[0006]
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2
However, the method of periodically calibrating in the field actually used as in Patent Document
2 described above is not suitable for facility equipment such as an intercom apparatus, and as in
Patent Document 3, a plurality of manufacturing processes are performed. In the method of
performing calibration by switching the range, a large-scale measuring device is required, and
the operation time is long.
[0007]
Thus, there is a need for a method of simply performing calibration of each microphone with
respect to an intercom apparatus that aims to prevent howling by canceling a speaker sound
from a transmission signal using a plurality of microphones.
[0008]
The present invention has been made in view of the above-mentioned problems, and its object is
to simplify the calibration of the microphone of the intercom apparatus which aims to prevent
howling by canceling the speaker sound from the speech signal using a plurality of microphones.
An object of the present invention is to provide a calibration method for intercom apparatus that
can be performed.
[0009]
According to the first aspect of the present invention, there is provided a sounding body for
emitting a sound by transmitted sound information, first and second microphones for collecting
sound and outputting a sound signal, first and second microphones, and a sounding body
Calculation processing using the delay time which is the transmission time of the sound wave
corresponding to the difference between each and the sound factor, and the coefficient calculated
from the amplitude ratio of each sound signal of the first and A signal processing unit that
generates an audio signal that cancels a sound emitted by a sounding body by applying to audio
signals output by the first and second microphones, and a coefficient that the signal processing
unit uses for arithmetic processing A calibration method for an intercom apparatus, wherein the
interphone apparatus is calibrated with respect to the storage unit, the interphone apparatus
comprising a sound generator, a first microphone, a second microphone, and a signal Logic unit,
a driving step of driving the sounding body at a specific frequency in a state where the storage
unit is incorporated, and the amplitude of each sound signal of the first and second microphones
for the sound from the sounding body driven at the specific frequency. An amplitude ratio data
deriving step of deriving data of the amplitude ratio, and the delay based on a phase of each
voice signal of the first and second microphones with respect to voice from a sounding body
driven at a specific frequency. A delay time data deriving step of deriving time data; an optimal
coefficient calculating step of calculating a coefficient used in the arithmetic processing by the
signal processing unit based on each of the derived delay time and the amplitude ratio data;
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Storing the calculated coefficient in a storage unit.
[0010]
According to the present invention, the intercom device is calibrated in a state where the
sounding body, the first and second microphones, and the signal processing unit are assembled,
and the sound emitted from the sounding body incorporated in the intercom device is used. Since
the coefficients used in the arithmetic processing for canceling the sound emitted by the
sounding body are calculated, there is no need to prepare a sounding body separately, and the
configuration of the calibration device can be simplified.
That is, by canceling the speaker sound from the transmission signal using a plurality of
microphones, calibration of the microphone of the intercom device can be easily performed to
achieve howling prevention. The driving step drives the sounding body at a specific frequency
sequentially selected from a plurality of frequencies, and the amplitude ratio data deriving step
and the delay time data deriving step process a specific frequency selected sequentially from a
plurality of frequencies. Each data of the said amplitude ratio and the said delay time is derived |
led-out.
[0011]
According to the present invention, since each data is derived in consideration of the frequency
characteristics of the audio signals of the first and second microphones, the accuracy of
calibration becomes high.
[0012]
In the invention of claim 3, according to claim 2, the optimum coefficient calculating step is
performed based on each data of the amplitude ratio and the delay time derived for each specific
frequency sequentially selected from a plurality of frequencies. A first coefficient used for the
arithmetic processing is calculated for each specific frequency, and a plurality of first coefficients
calculated for each specific frequency are averaged to calculate a second coefficient used for the
arithmetic processing. The storing step stores the second coefficient in the storing unit.
[0013]
According to the present invention, since only the second coefficient obtained by averaging the
first coefficient is calculated, the time required for calibration can be shortened.
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[0014]
According to a fourth aspect of the present invention, in the third aspect, the storage step stores
the second coefficient in the storage unit if the second coefficient is within a predetermined
range, and the second coefficient is in a predetermined range. If it is outside, the second
coefficient is not stored in the storage unit.
[0015]
According to the present invention, since the second coefficient which is an incorrect value is not
stored in the storage unit, the reliability of the intercom device can be secured.
[0016]
In the invention of claim 5, according to any one of claims 1 to 4, the signal processing unit
includes A / D conversion means for A / D converting each audio signal of the first and second
microphones, and the storage unit Stores in advance the sampling time of the A / D conversion
means, and the optimum coefficient calculating step performs the arithmetic processing based on
the sampling time read out from the storage unit in addition to each data of the delay time and
the amplitude ratio. Calculating the coefficients used for
[0017]
According to the present invention, calibration can be performed with respect to a plurality of
types of intercom devices (for example, a plurality of models having different distances between
the first and second microphones), and versatility is enhanced.
[0018]
As described above, according to the present invention, by canceling the speaker sound from the
transmission signal using a plurality of microphones, there is an effect that calibration of the
microphones of the intercom apparatus can be easily performed to prevent howling.
[0019]
Hereinafter, embodiments of the present invention will be described based on the drawings.
[0020]
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5
Embodiment 1 An intercom device A according to the present embodiment is shown in FIGS. 13
to 15, and includes an apparatus body A1 forming an outer shell of the apparatus, a speaker
module A2 attached to the apparatus body A1, an audio processing module A3, and a
microphone. M1 (first microphone), microphone M2 (second microphone), and various button
switches SW1 to SW3 are attached to the front of the embedded box 90 at an appropriate place
in the building.
Then, the information line Ls wired via the box 90 is connected, and functions as an intercom
apparatus capable of making a two-way call between the rooms via the information line Ls.
The power supply of the intercom device A may be supplied from an outlet provided near the
installation site, or may be supplied via the information line Ls.
Note that, in FIG. 14, part of the configuration requirements of the speaker SP described later
(yoke 20, permanent magnet 22 and the like) is omitted.
[0021]
The apparatus main body A1 is formed into a substantially box shape with the back surface and
upper and lower surfaces opened by molding ABS (Acrylonitrile-Butadiene-Styrene resin) or PCABS (PolyCarbonate-Acrylonitrile-Butadiene-Styreneresin), and from the lower side thereof A
locking piece 101 extending rearward and having a locking claw on the outside is locked to a
locking projection (not shown) of the box 90 and further not shown via a notch 102 formed
substantially at the center of the upper and lower ends. The main assembly A1 is attached to the
box 90 by screwing the attachment screw into a screw hole (not shown) of the box 90.
The apparatus body A1 is provided with mounting holes 103 at its upper and lower ends, and is
fixed to a structure such as a wall surface by inserting a screw into the mounting hole 103.
[0022]
Further, the rear surface of the box-like speaker module A2 is locked by the locking pieces 104,
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105 extending backward from the upper side of the device body A1 and having locking claws
inside, whereby the speaker module A2 is A plurality of sound holes 7 are provided on the front
of the device body A1 attached to the inside of the device body A1 and opposed to the speaker
module A2 for passing sound from the speaker SP included in the speaker module A2.
The sound holes 7 have a plurality of sound holes 7 formed in a substantially circular shape
substantially at the center of the front upper portion of the apparatus body A1, and further, a
plurality of dimples 7a formed by dimple processing are formed around the sound holes.
[0023]
Further, the rear surface of the box-like voice processing module A3 is locked by the locking
piece 106 extending backward from the lower side of the device body A1 and having locking
claws inside, so that the voice processing module A3 is It is attached to the inside of the device
body A1.
Openings 110, 111, and 112 are provided on the front of the device main body A1 to which the
voice processing module A3 faces, and projections respectively formed on the back of the call
button SW1, alarm stop button SW2, and indoor call button SW3. A pair of mounting holes 103
which are inserted from the front into the openings 110, 111 and 112 and attached to the front
of the voice processing module A3 and formed in the lower part of the device main body A1 are
covered by the call button SW1.
[0024]
A plate A4 having a plurality of recesses 7a is attached from the front by locking in the upper
part of the device body A1, and the pair of attachment holes 103 formed in the upper part of the
device body A1 is covered by the plate A4.
[0025]
Further, as shown in FIGS. 16 (a) to 16 (c), the Z1-Z1 cross section of FIG. 16 (b) is shown in FIG.
16 (a), and the Z2-Z2 cross section of FIG. 16 (c), on the inner surface of the device body A1 to
which the speaker module A2 faces, a first cylindrical boss 71 is protruded substantially at the
center of a plurality of sound holes 7 formed in a plurality of substantially circular shapes. The
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7
second cylindrical boss 72 is provided on the side of the sound hole 7 (where it does not face the
sound hole 7), and the circular recess 71a formed on the end face of the boss 71 in the axial
direction The microphone M1 is attached, and the microphone M2 is attached to the circular
recess 72a formed on the axial end face of the boss 72, so that the microphones M1 and M2 can
be easily positioned.
Here, the microphones M1 and M2 have a disk-like outer shell, and a rubber cap G is provided on
the outer peripheral surface thereof. When the microphones M1 and M2 are press-fit into the
recesses 71a and 72a, the recess 71a is formed by the elasticity of the rubber cap G. , 72a, so
that assembly is facilitated without using a fixing means such as an adhesive.
Alternatively, if an elastic elastomer is formed on the inner side surfaces of the concave portions
71a and 72a, the microphones M1 and M2 need not be provided with the rubber cap G, and the
microphones M1 and M2 can be easily fixed in the concave portions 71a and 72a. it can.
Furthermore, the entire bosses 71 and 72 may be formed of an elastic elastomer.
[0026]
Furthermore, a pair of wires W1 and W2 for outputting audio signals are respectively derived
from the microphones M1 and M2, and the wires W1 and W2 are disposed in the groove
portions 71b and 72b formed downward from the concave portions 71a and 72a. Thus, the
signal processing module M3 is drawn into the sound processing module M3 to prevent
interference with the speaker module A2 and secure a wiring path.
The microphones M1 and M2 are configured by back electret type electret condenser
microphones.
[0027]
A speaker module A2, an audio processing module A3 and microphones M1 and M2 are attached
to the inner surface of the apparatus main body A1, and a power supply circuit (not shown)
provided in the box 90 is supplied from outside. AC is converted into an operation power supply
of an internal circuit composed of a stable DC voltage and supplied to each part.
04-05-2019
8
[0028]
The speaker module A2 is attached to the inner surface of the device main body A1 so as to face
the sound hole 7 and, as shown in FIG. 14, has an opening on the rear surface and covers the
resin molded body A21 and the opening of the body A21. A module main body A20 of 40 mm
wide × 30 mm high × 8 mm thick is configured with a flat resin molded cover A22 to be
provided, and the speaker SP is provided in the module main body A20.
[0029]
As shown in FIG. 13, the speaker SP has a thickness of about 0.8 mm, such as a cold-rolled steel
plate (SPCC, SPCEN) or electromagnetic soft iron (SUY) formed on the bottom of a circular recess
11 provided on the front of the body A21. An annular yoke 20 formed of an iron-based material
and a cylindrical support 21 extending forward from the outer peripheral edge of the yoke 20
are integrally formed in the recess 11 of the body A21. .
[0030]
The cylindrical permanent magnet 22 (for example, residual magnetic flux density 1.39T to
1.43T) formed of NdFeB is disposed in the opening 20a in the annular ring of the yoke 20, and
the edge on the outer peripheral side of the dome-shaped diaphragm 23 The portion is adhered
to the step surface 21 a of the support 21.
[0031]
The diaphragm 23 is formed of a thermoplastic plastic (for example, a thickness of 12 μm to 50
μm) such as PET (PolyEthylene Terephtalate) or PEI (Polyetherimide).
A cylindrical bobbin 24 is fixed to the back surface of the diaphragm 23, and is formed by
winding a polyurethane copper wire (for example, φ 0.05 mm) around a kraft paper tube at the
rear end of the bobbin 24. A voice coil 25 is provided.
The bobbin 24 and the voice coil 25 have a cylindrical permanent magnet 22 disposed inside, are
provided opposite to the yoke 20, and freely move in the front-rear direction in the vicinity of the
yoke 20.
04-05-2019
9
[0032]
Further, on the front surface of the body A21, as shown in FIG. 14, a terminal plate 30 is
disposed on the side of the speaker SP, and a pair of terminal portions provided on the terminal
plate 30 has a pair of voice coils 25 of the speaker SP. The lead wires are connected by soldering,
and the output wiring from the audio processing module A3 is also connected by soldering.
[0033]
Then, when an audio signal is input to the polyurethane copper wire of the voice coil 25, an
electromagnetic force is generated in the voice coil 25 by the current of the audio signal and the
magnetic field of the permanent magnet 22, so the bobbin 24 is accompanied by the diaphragm
23. It is vibrated in the back and forth direction.
At this time, the diaphragm 23 emits a sound according to the audio signal.
That is, a dynamic speaker SP is configured.
[0034]
When the speaker SP is attached to the module body A20, a rear air chamber Br is formed, which
is a space surrounded by the inner side and the inner side of the rear surface of the module body
A20 and the rear surface side (yoke 20 side) of the speaker SP.
In the rear air chamber Br, a space in which the diaphragm 23 of the speaker SP is in close
contact with the step surface 21a of the support 21 and the body A21 of the module body A20 is
in close contact with the cover A22. become.
[0035]
When the speaker module A2 is attached to the device body A1, as shown in FIG. 13, the speaker
module A2 is surrounded by the inner surface of the device body A1 and the surface side
(diaphragm 23 side) of the speaker SP and is insulated from the rear air chamber Br. A front air
chamber Bf is formed, and the microphone M1 attached to the boss 71 on the back of the
apparatus main body A1 is disposed in the front air chamber Bf with its sound collecting surface
04-05-2019
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facing substantially the center of the diaphragm 23. , Has high directivity to the sound from the
speaker SP.
[0036]
When the boss 72 on the back surface of the device body A1 is fitted in the recess 31 provided
on the front surface of the body A21 of the speaker module A2 and the boss 72 is fitted in the
recess 31, the speaker module A2 is assembled to the device body A1. Positioning can be easily
performed, and assembly work can be smoothly performed.
Further, the microphone M2 attached to the boss 72 communicates with the outside through the
sound collecting hole 8 formed on the front surface of the apparatus main body A1 with its
sound collecting surface transmitted through the sound collecting hole 8 The voice from the
speaker located in front of the intercom device A has high directivity.
[0037]
Next, as shown in FIG. 17, the audio processing module A3 is composed of an IC including a
communication unit 81, audio switch units 82 and 83, an amplification unit 84, a signal
processing unit 85, and a storage unit 86, and other rooms A voice signal transmitted from the
intercom device A installed in the like via the information line Ls is received by the
communication unit 81, amplified by the amplification unit 84 through the voice switch unit 82,
and then output from the speaker SP Be done.
In addition, by operating the call button SW1, the communication becomes possible, and each
audio signal input from the microphone M1 and the microphone M2 is subjected to signal
processing to be described later by the signal processing unit 85 and then passes through the
audio switch unit 83. The communication unit 81 transmits the interphone device A installed in
another room or the like through the information line Ls.
Further, the alarm stop button SW2 is operated when stopping the notification by the alarm
signal received from the other terminal device through the information line Ls, and the indoor
call button SW3 is used to call the intercom device A installed in the other room To operate.
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[0038]
Then, assuming that distances from the center of the diaphragm 23 of the speaker SP to the
centers of the sound collecting surfaces of the microphones M1 and M2 are X1 and X2,
respectively, X1 <X2 and in this embodiment, the audio output of the speaker SP is a microphone
The following configuration is provided to prevent howling caused by M1 and M2 picking up.
[0039]
First, as shown in FIG. 18, the signal processing unit 85 housed in the audio processing module
A3 includes A / D conversion circuits 85a and 85b for A / D converting each audio signal of the
microphones M1 and M2, and A / D conversion circuit A delay circuit 85c for delaying the
output of the D conversion circuit 85a, an amplitude adjustment circuit 85d for adjusting the
amplitude of the output of the delay circuit 85c, and an amplification circuit 85d output from the
audio signal of the microphone M2 output from the A / D conversion circuit 85b. And a
subtracting circuit 85e that subtracts the audio signal of the microphone M1.
[0040]
FIGS. 19 to 22 show sound signal waveforms of respective portions of the signal processing unit
85 when the sound from the speaker SP is collected by the microphones M1 and M2,
respectively.
First, assuming that distances from the center of the diaphragm 23 of the speaker SP to the
centers of the sound collecting surfaces of the microphones M1 and M2 are X1 and X2,
respectively, X1 <X2.
Therefore, when the sound from the speaker SP is picked up by the microphones M1 and M2, the
amplitude of the output of the microphone M2 is larger than that of the microphone M1
depending on the distance between the speaker SP and the microphones M1 and M2 and the
sensitivity of the microphones M1 and M2. The delay time of the sound wave [Td = (X2-X1) / Cv]
(Cv is the velocity of sound) corresponding to the difference (X2-X1) of the distance between the
two microphones M1 and M2 and the speaker SP is small, and the output of the microphone M2
is The phase is behind.
[0041]
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Therefore, the output Y21 of the microphone M2 after A / D conversion has a smaller amplitude
than the output Y11 of the microphone M1 after A / D conversion, and the output Y21 of the
microphone M2 is phased by the delay time Td from the output Y11 of the microphone M1. Is
late.
(Refer FIG. 19 (a) (b)).
[0042]
The delay circuit 85c is formed of a time delay element or a CR phase delay circuit, and delays
the output of the microphone M1 closer to the speaker SP by the delay time Td, thereby
outputting the outputs Y12 and A of the delay circuit 85c. The phases are made to coincide with
the output Y21 of the / D conversion circuit 85b (see FIGS. 20 (a) and 20 (b)).
[0043]
Then, the amplitude adjustment circuit 85d generates the output Y13 obtained by adjusting the
amplitude of the output Y12. At this time, the amplitude adjustment is performed to match the
amplitudes of the output Y13 and the output Y21 with respect to the sound from the speaker SP.
The output levels of both microphones M1 and M2 with respect to the sound from the speaker
SP are made to match (see FIGS. 21 (a) and 21 (b)).
In the present embodiment, the amplification factor of the amplitude adjustment circuit 85d
connected to the microphone M1 close to the speaker SP is less than one.
[0044]
The audio component from the speaker SP included in the output Y13 and the audio component
from the speaker SP included in the output Y21 have the same amplitude and the same phase by
the above delay processing and amplitude adjustment processing, and the subtraction circuit 85e
outputs By subtracting Y13 from Y21, an output Ya in which the sound from the speaker SP is
canceled is generated (see FIG. 22).
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That is, at the output Ya, the sound component from the speaker SP is reduced.
[0045]
Further, for the sound from the speaker SP, the amplitude of the output Y11 of the microphone
M1 arranged with the sound collection surface facing the diaphragm 23 of the speaker SP is the
microphone with the sound collection surface arranged toward the speaker H The amplitude of
the output Y21 of the microphone M2 is larger than the amplitude of the output Y11 of the
microphone M1 for the voice emitted by the speaker H in front of the microphones M1 and M2
while the amplitude is larger than the amplitude of the output Y21 of M2. Become. Furthermore,
since the amplification factor of the amplitude adjustment circuit 85d is less than 1, the speech
component from the speaker H included in the output Y21 becomes larger than the speech
component from the speaker H included in the output Y13. That is, the amplitude difference
between the speech component from the speaker H included in the output Y13 and the speech
component from the speaker H included in the output Y21 becomes large, and the output Ya is
obtained even if the subtraction processing is performed by the subtraction circuit 85e. , The
signal corresponding to the speech emitted by the speaker H remains with sufficient amplitude.
[0046]
As described above, at the output Ya of the subtraction circuit 85e, the speech component from
the speaker SP is reduced, and the speech component emitted by the speaker H ahead of the
intercom device A remains, and at the output Ya, the speaker H who wants to leave The relative
difference between the sound component from the speaker SP and the sound component from
the speaker SP to be reduced is large. That is, even when the voice from the speaker H and the
voice from the speaker SP are simultaneously generated, only the voice component from the
speaker SP is reduced while maintaining sufficient amplitude for the voice component from the
speaker H. Therefore, it is possible to prevent the howling caused by the microphones M1 and
M2 picking up the audio output of the speaker SP.
[0047]
However, the delay time Td for the sound from the speaker SP and the amplitudes of the sound
signals Y11 and Y21 are under ideal conditions (for example, point sound source, housing
sealing structure, variation in CR of circuit configuration, etc.) Although it is constant with
respect to the frequency without depending on the frequency of the sound emitted by the SP, it is
04-05-2019
14
not dependent on the frequency of the sound emitted by the speaker SP because it is not under
ideal conditions in practice.
[0048]
Therefore, since the delay time Td between the audio signals Y11 and Y21 with respect to the
sound from the speaker SP and the amplitude of the audio signals Y11 and Y21 have frequency
characteristics, the arithmetic processing in the delay circuit 85c, the amplitude adjustment
circuit 85d and the subtraction circuit 85e Is expressed by an algorithm using a plurality of
circuit blocks 85i (i = 1, 2,..., N) corresponding to voice signals of different frequencies as shown
in FIG. Is considered.
[0049]
The mth digital data of the output Y11 of the A / D conversion circuit 85a connected to the
microphone M1 is S (m), and the mth digital data of the output Y21 of the A / D conversion
circuit 85b connected to the microphone M2 is , Each circuit block 85i includes an amplification
unit 90i for amplifying the mth digital data S (m) of the microphone M1, and the m−1th digital
data S (m−) of the microphone M1 via the delay element 91i. 1), an amplification unit 92i that
amplifies the outputs of the amplification units 90i and 91i, an amplification unit 94i that
amplifies the output of the addition unit 93i, and the mth digital data R (m of the microphone M2
And a subtraction unit 95i that subtracts the output of the amplification unit 94i from.
The averaging processor 850 adds the outputs of the subtractors 95i of the circuit blocks 85i
and divides the sum by n, thereby generating an output Ya obtained by averaging the outputs of
the circuit blocks 85i. .
[0050]
The amplification factor Bi1 of the amplification unit 90i, the amplification factor Bi2 of the
amplification unit 92i, and the amplification factor Ci of the amplification unit 94i cancel the
speaker sound most with respect to the audio signal of the frequency corresponding to the
circuit block 85i. It can be set to a value that can.
That is, the speaker sound of each output of the circuit block 85i is canceled for each different
04-05-2019
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frequency.
[0051]
The output Ya generated by the averaging processing unit 850 is represented using a total of 3n
coefficients (first coefficients) such as Bi1, Bi2, and Ci as in [Equation 1].
[0052]
[0053]
Further, since [Equation 1] can be summarized as [Equation 2], the output Ya has two coefficients
(K1 and K2 obtained by averaging Bi1, Bi2 and Ci (first coefficient) It can be expressed using only
a factor of 2.
[0054]
[0055]
Therefore, according to [Equation 2], as shown in FIG. 24, the arithmetic processing in delay
circuit 85c and amplitude adjustment circuit 85d is amplification section 96 which amplifies mth
digital data S (m) of microphone M1 by amplification factor K1. The amplification unit 98
amplifies the (m−1) th digital data S (m−1) of the microphone M1 by the amplification factor
K2 through the delay element 97, and the addition unit 99 adds the outputs of the amplification
units 96 and 98. The subtraction circuit 85e subtracts the output of the addition unit 99 from
the m-th digital data R (m) of the microphone M2 to generate an output Ya.
In this algorithm, only two coefficients, that is, the amplification factor K1 of the amplification
unit 96 and the amplification factor K2 of the amplification unit 98 may be set, and the algorithm
of signal processing can be simplified as compared with FIG.
[0056]
In this embodiment, the algorithm of FIG. 24 is used as the delay circuit 85c and the amplitude
adjustment circuit 85d, and the two coefficients [amplification factors K1 and K2] are stored in
the storage unit 86 such as ROM, The adjustment circuit 85d performs arithmetic processing
04-05-2019
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based on the coefficients [amplification factors K1, K2] stored in the storage unit 86. These
coefficients are stored in the storage unit 86 by calibration described later.
[0057]
Next, in the voice switch units 82 and 83 (see FIG. 17), howling is further prevented by
performing the following processing.
The voice switch unit 82 is disposed on the transmission line of the received signal, and the voice
switch unit 83 is disposed on the transmission line of the signal to be transmitted. The voice
switch units 82 and 83 compare the levels of the input signals with each other. The voice switch
unit with the smaller level of the input signal increases the transmission loss on the transmission
line by the variable loss means provided internally.
Therefore, of the received signal and the signal to be transmitted, a signal with a smaller level is
attenuated, and the howling margin is further increased, thereby further preventing howling.
[0058]
Next, the coefficients [amplification factors K1 and K2] used when the delay circuit 85c and the
amplitude adjustment circuit 85d of the signal processing unit 85 perform arithmetic processing
are set, and these coefficients [amplification factors K1 and K2] are stored in the storage unit 86.
The calibration stored in will be described.
[0059]
The calibration device B is installed in the anechoic chamber 200 installed before the packing
process of the manufacturing line of the intercom device A as shown in FIG. 2, and the anechoic
chamber 200 has a width of 1.5 × It has a rectangular box shape of depth 1.5 × height 2.0 (m),
and as shown in FIG. 3, an opening 201 for allowing the conveyor CV to penetrate is provided on
both side surfaces.
The interior of the anechoic chamber 200 is suitable for calibration because external noise, noise
and the like are shielded.
04-05-2019
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[0060]
The inside of the anechoic chamber 200 is covered with the sound absorbing material 202 as
shown in FIG. 4 and the personal computer 211 and the intercom device A under calibration are
fixed on the work desk 210 installed near the conveyor CV. The fixing jig 212 is disposed, and
the operator 290 sets the intercom apparatus A before calibration from the conveyor CV to the
fixing jig 212.
[0061]
As shown in FIG. 5, the personal computer 211 is installed with application software 211a
(hereinafter referred to as measurement analysis software 211a) as a measurement analysis tool,
and also includes an A / D conversion unit 211b, a memory writer 211c, a RAM, etc. A storage
unit 211d and the like are provided to constitute measurement calculation means and coefficient
writing means.
[0062]
Then, after the operator 290 operates the personal computer 211 to activate the measurement
analysis software 211a, calibration is performed according to the flowchart of FIG.
First, the operator 290 connects the output port 211e of the personal computer 211 to the
(voice coil 25 of) the speaker SP of the intercom device A via the speaker driving amplifier 220,
and A / D converts the outputs of the microphones M1 and M2. The input / output terminal of
the memory writer 211c is connected to the input / output terminal of the storage unit 86 in the
audio processing module A3 (S1).
The speaker driving amplifier 220 includes an amplifier control unit 220a and an amplification
unit 220b. The amplifier control unit 220a controls the operation of the amplification unit 220b
according to an instruction from the personal computer 211.
A connector (not shown) may be provided in the intercom device A, and the wires from the
personal computer 211 and the amplifier 220 may be connector-connected to simplify the
connection work.
04-05-2019
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[0063]
Next, the measurement analysis software 211a that has started calibration by the operation of
the worker 290 drives the speaker SP via the amplifier 220, and the A / D conversion unit 211b
samples the speaker sound collected by the microphones M1 and M2. Then, the drive process for
A / D conversion is started, and at this time, the frequency of the sine wave input to the speaker
SP is sequentially selected, and the speaker SP is sequentially driven at a plurality of frequencies.
[0064]
In addition, data of the sampling time of the A / D conversion circuits 85a and 85b are stored in
advance in the storage unit 86 of the intercom device A, and the measurement analysis software
211a starts sampling time from the storage unit 86 of the intercom device A at startup. The data
of is read out and set as a sampling time Ts of A / D conversion performed by the A / D
conversion unit 211b.
That is, by matching the sampling time in the intercom device A with the sampling in the
personal computer 211, the calibration accuracy is improved.
[0065]
First, the speaker SP is driven with a sine wave of 500 Hz (S2), and the A / D converter 211b A /
D converts the speaker sound collected by the microphones M1 and M2 at that time, and the
measurement analysis software 211a performs A / D conversion. The signal after D conversion is
stored in the storage unit 211d as an audio signal Ys1 of the microphone M1 and an audio signal
Ys2 of the microphone M2 (S3).
[0066]
Next, the speaker SP is driven with a 1 KHz sine wave (S4), and the A / D converter 211b A / D
converts the speaker sound collected by the microphones M1 and M2 at that time, and the
measurement analysis software 211a The signal after the / D conversion is stored in the storage
unit 211d as an audio signal Ys1 of the microphone M1 and an audio signal Ys2 of the
microphone M2 (S5).
[0067]
Next, the speaker SP is driven with a 2 KHz sine wave (S6), and the A / D converter 211b A / D
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converts the speaker sound collected by the microphones M1 and M2 at that time, and the
measurement analysis software 211a The signal after the / D conversion is stored in the storage
unit 211d as an audio signal Ys1 of the microphone M1 and an audio signal Ys2 of the
microphone M2 (S7).
[0068]
Then, the measurement analysis software 211a starts the amplitude ratio data derivation process
and the delay time data derivation process based on the audio signals Ys1 and Ys2 obtained by
collecting the sine waves of 500 Hz, 1 KHz, and 2 KHz.
[0069]
The amplitude ratio data deriving step (S8) will be described below.
First, assuming that the circuit block 851 in the algorithm before simplification shown in FIG. 23
corresponds to the speaker sound of 500 Hz, the circuit block 852 corresponds to the speaker
sound of 1 KHz, and the circuit block 853 corresponds to the speaker sound of 2 KHz. Think.
[0070]
And audio signal Ys1 which microphone M1 which collected the speaker sound of 500 Hz
outputs is peak value V11 (peak to peak) like Fig.6 (a), and microphone M2 which collected the
speaker sound of 500 Hz is The audio signal Ys2 to be output is the peak value V21 (peak to
peak) as shown in FIG. 6B, and the amplitude ratio data at this time is [V11 / V21], and this
amplitude ratio data [V11 / V21] is And the amplification factor C1 of the amplification unit 941.
That is, it is represented by C1 = V11 / V21.
[0071]
The audio signal Ys1 output by the microphone M1 that collects the 1 KHz speaker sound is a
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20
peak value V12 (peak to peak) as shown in FIG. 7A, and is output by the microphone M2
collecting the 1 KHz speaker sound. The audio signal Ys2 has a peak value V22 (peak to peak) as
shown in FIG. 7B, the amplitude ratio data at this time is [V12 / V22], and this amplitude ratio
data [V12 / V22] is amplified It becomes equal to the amplification factor C2 of the part 942.
That is, C2 = V12 / V22.
[0072]
The audio signal Ys1 output from the microphone M1 that has collected the 2 KHz speaker
sound is a peak value V13 (peak to peak) as shown in FIG. 8A, and is output by the microphone
M2 that collects the 2 KHz speaker sound. The audio signal Ys2 has a peak value V23 (peak to
peak) as shown in FIG. 8B, and the amplitude ratio data at this time is [V13 / V23], and this
amplitude ratio data [V13 / V23] is amplified It becomes equal to the amplification factor C3 of
the part 943.
That is, C3 = V13 / V23.
[0073]
Next, from the audio signals Y11 and Y21 of the microphones M1 and M2 that collected the 500
Hz, 1 KHz, and 2 KHz speaker sounds, the delay time Td at which the phase of the output of the
microphone M2 lags the output of the microphone M1 It derives for each frequency (S9).
[0074]
First, as for the audio signals Ys1 and Ys2 output from the microphones M1 and M2 with respect
to the speaker sound of 500 Hz, the audio signal Ys2 of the microphone M2 is the audio signal
Ys1 of the microphone M1 as shown in FIGS. The delay time is delayed by Td1 compared to.
[0075]
As for each audio signal Ys1 and Ys2 which microphones M1 and M2 output to the speaker
sound of 1 KHz, as shown in FIGS. 10 (a) and 10 (b), the audio signal Ys2 of microphone M2 is
compared with the audio signal Ys1 of microphone M1. Delay time Td2.
04-05-2019
21
[0076]
As for the audio signals Ys1 and Ys2 output from the microphones M1 and M2 with respect to
the speaker sound of 2 KHz, the audio signal Ys2 of the microphone M2 is compared with the
audio signal Ys1 of the microphone M1 as shown in FIGS. Delay time Td3.
[0077]
Next, based on each amplitude ratio data [C1 = V11 / V21, C2 = V12 / V22, C3 = V13 / V23] at
500 Hz, 1 KHz and 2 KHz, and each delay time data [Td1, Td2, Td3] The optimal coefficient
deriving step is started to derive the coefficient [amplification factor K1, K2] optimal for the
calculation process for canceling the speaker sound (S10).
Here, the delay times Td1, Td2, and Td3 at 500 Hz, 1 KHz, and 2 KHz will be described below as
having a relationship of Td1 <Td2 <Td3.
[0078]
First, the amplification factor Bi1 of the amplification unit 90i of the circuit block 85i
corresponding to the frequency of the shortest delay time Td is set to “1”, and the
amplification factor Bi2 of the amplification unit 92i is set to “0”.
Here, since the delay time Td1 is the shortest, the amplification factor B11 of the amplification
unit 901 of the circuit block 851 corresponding to 500 Hz is “1”, and the amplification factor
B12 of the amplification unit 921 is “0”.
[0079]
The amplification factors Bi1 and Bi2 of the circuit block 85i of the circuit block corresponding
to the other frequency are obtained by the difference between the delay time at each frequency
and the shortest delay time.
For example, in FIG. 12, the A / D conversion unit 211b performs A / D conversion at sampling
04-05-2019
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time Ts, and the plot P1 represents the mth digital data S of the output of the microphone M1
corresponding to the shortest delay time frequency. (M) is shown, and a plot P2 shows the (m-1)
-th digital data S (m-1) of the output of the microphone M1 corresponding to the frequency of
the shortest delay time.
Then, assuming this difference ΔTd between the shortest delay time and the delay time at
another frequency, the plot P3 delayed by the difference ΔTd from the plot P1 corresponds to
the other frequency on the straight line connecting the plots P1 and P2. It is considered that the
mth digital data of the output of the microphone M1 is present.
[0080]
Thus, [Equation 3] is derived from FIG. 12, and amplification factors Bi1 and Bi2 are expressed as
[Equation 4].
[0081]
[0082]
[0083]
Therefore, the amplification factor B21 of the amplification unit 902 and the amplification factor
B22 of the amplification unit 922 of the circuit block 852 corresponding to the frequency 1 KHz
are represented by [Equation 5].
[0084]
[0085]
The amplification factor B31 of the amplification unit 903 of the circuit block 853 corresponding
to the frequency 2 KHz and the amplification factor B32 of the amplification unit 923 are
expressed by [Equation 6].
[0086]
04-05-2019
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[0087]
Then, since [Equation 7] is obtained from [Equation 2], the amplification factor Ci of the
amplification unit 94i, the amplification factor Bi1 of the amplification unit 90i, and the
amplification unit 92i in the circuit block 85i corresponding to each frequency obtained as
described above. The amplification factor Bi2 is substituted into [Equation 7] and averaged, and
the coefficient [amplification factor K1, K2 used when the delay circuit 85c and the amplitude
adjustment circuit 85d of the signal processing unit 85 perform arithmetic processing (algorithm
in FIG. ] Is calculated (S10).
[0088]
[0089]
As described above, since the coefficients [amplification factors K1 and K2] are calculated in
consideration of the frequency characteristics of the audio signals of the microphones M1 and
M2, the calibration accuracy is enhanced.
Further, since only two coefficients (second coefficients) of K1 and K2 obtained by averaging Bi1,
Bi2 and Ci (first coefficient) are calculated, the time required for calibration can be shortened.
[0090]
Here, the measurement analysis software 211a reads data of the sampling time from the storage
unit 86 of the intercom device A at startup and sets it as a sampling time Ts of A / D conversion
performed by the A / D conversion unit 211b.
Therefore, the above calibration can be performed on a plurality of types of intercom devices A
(for example, a plurality of models having different distances between the microphones M1 and
M2), and the versatility is enhanced.
[0091]
04-05-2019
24
Next, the measurement analysis software 211a of the personal computer 211 determines
whether or not the coefficient [amplification factor K1, K2] calculated in S10 is within a preset
range, and the factor [amplification factor K1, K2] is If it falls within the range, the calculated
coefficient [amplification factor K1, K2] is stored in the storage unit 86 of the intercom device A
by the memory writer 211c as an appropriate value.
If the factor [amplification factor K1, K2] does not fall within the range, the calculated factor
[amplification factor K1, K2] is discarded as an inappropriate value, and the inappropriate factor
[amplification factor K1, K2] Are stored in the storage unit 86, and the reliability of the intercom
device A is secured (S11).
[0092]
Then, the operator 290 in the anechoic chamber 200 calculates the coefficient [amplification
factor K1, K2] at an appropriate value and stores the intercom device A stored in the storage unit
86 on the conveyor CV again Take it out of 200.
The operation of placing the intercom device A on the conveyor CV may be performed by a robot
machine (not shown) to reduce the number of man-hours.
On the other hand, the interphone device A 'for which the coefficient [amplification factors K1
and K2] is calculated at an inappropriate value and the calibration becomes defective is stored in
the tray 230 in the anechoic chamber 200 (see FIG. 4).
Then, whether or not there is a defect in the microphones M1 and M2, a disconnection in the
wiring, or the like is checked to replace the part that has caused the calibration failure or repair
is performed, and then calibration is performed again.
[0093]
The intercom device A for which the calibration has been completed is transported from the
anechoic chamber 200 to the packing process by the conveyor C, packaged by the operator 300
in the cardboard box 301, and shipped (see FIG. 2).
04-05-2019
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[0094]
As described above, the intercom device A is calibrated with the speaker SP, the microphones M1
and M2, and the voice processing module assembled in the device main body A1, and the sound
emitted from the speaker SP incorporated in the interphone device A is used. Since the
coefficients [amplification factors K1 and K2] used for the calculation process for canceling the
speaker sound are calculated, there is no need to prepare a sound generator such as a separate
speaker, and the configuration of the calibration device can be simplified.
[0095]
In the present embodiment, voice signals are exchanged between the intercom devices A using a
wired communication method via the information line Ls. However, by providing the intercom
device A with known wireless communication means, wireless communication can be performed.
Transmission and reception of an audio signal may be performed by a communication method.
[0096]
Second Embodiment In the first embodiment, calibration is performed by connecting the external
personal computer 211 and the amplifier 220 having the measurement analysis software 211a
installed to the intercom device A, but the intercom device A of this embodiment is As shown in
FIG. 25, a microcomputer 240 (hereinafter referred to as the microcomputer 240) is stored
inside.
As shown in FIG. 26, the microcomputer 240 is installed with application software 240a
(hereinafter referred to as measurement analysis software 240a) as a measurement analysis tool,
and has a storage unit 240b such as a RAM, etc. Communication is possible via a terminal 250
such as a personal computer and the like via the external interface 260.
Further, the audio processing module A3 includes the respective functions of the amplifier 220,
the A / D conversion means 211b, and the memory writer 211c of the first embodiment, and the
microcomputer 240 and the audio processing module A3 constitute measurement calculation
means and coefficient writing means. .
[0097]
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Then, after the operator 290 operates the terminal 250 to access the microcomputer 240 and
activates the measurement analysis software 240a, the measurement analysis software 240a
performs calibration in accordance with the flowchart of FIG.
First, the measurement analysis software 240 a drives the speaker SP, and performs processing
in order of test mode 1 (S 21) and test mode 2 (S 22).
[0098]
First, in the test mode 1, the measurement analysis software 240a validates the input from the
microphone M1 and invalidates the input from the microphone M2, and then the speaker SP is
applied in the order of 500 Hz sine wave, 1 KHz sine wave, and 2 KHz sine wave. The voice
processing module A3 sequentially A / D converts the speaker sound that the microphone M1
collects at each frequency, and the measurement analysis software 240a converts the signal of
each frequency after A / D conversion to the voice signal Ys1 of the microphone M1. Are stored
in the storage unit 240b.
[0099]
Next, in test mode 2, the measurement analysis software 240a disables the input from the
microphone M1 and enables the input from the microphone M2, and then the speaker SP in the
order of 500 Hz sine wave, 1 KHz sine wave, 2 KHz sine wave. The audio processing module A3
sequentially A / D converts the speaker sound collected by the microphone M2 at each
frequency, and the measurement analysis software 240a converts the signal of each frequency
after the A / D conversion into an audio signal of the microphone M2. It is stored in the storage
unit 240b as Ys2.
[0100]
Then, the measurement analysis software 240a performs an amplitude ratio data derivation step
(S23) and a delay time data derivation step (S24) based on the audio signals Ys1 and Ys2
obtained by collecting the sine waves of 500 Hz, 1 KHz and 2 KHz. Then, based on each
amplitude ratio data [C1 = V11 / V21, C2 = V12 / V22, C3 = V13 / V23] at 500 Hz, 1 KHz and 2
KHz, and each delay time data [Td1, Td2, Td3] An optimum coefficient deriving step is carried
out to derive coefficients [amplification factors K1, K2] which are optimum for the arithmetic
processing for canceling the speaker sound (S25).
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The amplitude ratio data derivation step (S23), the delay time data derivation step (S24), and the
optimum coefficient derivation step (S25) are the amplitude ratio data derivation step (S8) of the
first embodiment, the delay time data derivation step (S9). And the optimal coefficient deriving
step (S10), and thus the detailed description will be omitted.
However, in this embodiment, since the sound signals of the microphones M1 and M2 are
collected separately in the test modes 1 and 2, the delay time in which the phase of the output of
the microphone M2 lags the output of the microphone M1 Td is derived on the basis of the
timing at which the voice of each frequency is output from the speaker SP.
[0101]
Next, the measurement analysis software 240a determines whether or not the coefficient
[amplification factor K1, K2] calculated in S25 is within a preset range, and the factor
[amplification factor K1, K2] is within the range If it is within the range, the calculated coefficient
[amplification factors K1, K2] is stored in the storage unit 86 of the intercom device A as an
appropriate value.
If the factor [amplification factor K1, K2] does not fall within the range, the calculated factor
[amplification factor K1, K2] is discarded as an inappropriate value, and the inappropriate factor
[amplification factor K1, K2] However, the reliability of the intercom device A is secured without
being stored in the storage unit 86 (S26).
[0102]
As described above, in the calibration method of the present embodiment, the intercom
apparatus A includes the microcomputer 240 installed with the measurement analysis software
240a, and the voice processing module A3 equipped with each function of an amplifier, A / D
conversion means, and memory writer. Therefore, there is no need to install measurement
analysis software on the external personal computer 211 or to prepare the amplifier 220, and
the configuration of the calibration device can be further simplified.
Furthermore, even after the intercom device A is installed, calibration can be easily performed
according to the installation situation simply by connecting the terminal device 250 capable of
04-05-2019
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communicating with the microcomputer 240 to the intercom device A.
The other configuration is the same as that of the first embodiment, and the description is
omitted.
[0103]
FIG. 2 is a diagram showing a flowchart of a calibration method for an intercom device of
Embodiment 1.
It is a figure which shows the outline | summary of a calibration installation same as the above. It
is a figure which shows the side of an anechoic room same as the above. It is the schematic
which shows the outline | summary of an anechoic chamber same as the above. It is a figure
which shows the structure of a calibration apparatus same as the above. (A) (b) It is a wave form
diagram of a 500 Hz audio | voice signal same as the above. (A) (b) It is a wave form diagram of
the audio | voice signal of 1 KHz same as the above. (A) (b) It is a wave form diagram of the 2
KHz audio | voice signal same as the above. (A) (b) It is a wave form diagram of a 500 Hz audio |
voice signal same as the above. (A) (b) It is a wave form diagram of the audio | voice signal of 1
KHz same as the above. (A) (b) It is a wave form diagram of the 2 KHz audio | voice signal same
as the above. It is a figure which shows the optimal coefficient calculation process same as the
above. It is sectional drawing which shows the intercom apparatus same as the above. It is a
disassembled perspective view which shows a part of interphone apparatus same as the above. It
is the perspective view which attached the intercom apparatus same as the above to the box. It is
a partially enlarged reverse view of the apparatus main body same as the above. It is a block
diagram of an audio processing module same as the above. It is a circuit block diagram of the
signal processing part same as the above. (A) (b) It is a signal waveform diagram of a signal
processing part same as the above. (A) (b) It is a signal waveform diagram of a signal processing
part same as the above. (A) (b) It is a signal waveform diagram of a signal processing part same
as the above. It is a signal waveform diagram of the signal processing part same as the above. It
is a figure which shows the algorithm before the simplification in a delay circuit same as the
above, and an amplitude adjustment circuit. It is a figure which shows the algorithm after
simplification in a delay circuit same as the above, and an amplitude adjustment circuit. FIG. 7 is
a diagram showing a configuration of a calibration device of Embodiment 2. It is a figure which
shows the structure of the microcomputer same as the above. It is a figure which shows the
flowchart of the calibration method for intercom apparatuses same as the above.
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Explanation of sign
[0104]
A Intercom system SP Speaker M1, M2 Microphone 211 PC 220 amplifier
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