JP2010081124A - Calibration method for intercom device - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a calibration method for an intercom device capable of easily executing the calibration of the microphone of the intercom device, for preventing howling by canceling speaker sound from transmission signals using a plurality of microphones. <P>SOLUTION: The calibration method for the intercom device includes: drive processes S2-S7 for driving the speaker SP of the intercom device A at a specified frequency; an amplitude ratio data deriving process S8 for deriving the data of the amplitude ratio of respective sound signals of microphones M1 and M2 for sound from the speaker SP; a delay time data deriving process (S9) for deriving the data of the delay time on the basis of the phase of the respective sound signals of the microphones M1 and M2 to the sound from the speaker SP; an optimum coefficient calculation process (S10) for calculating a coefficient optimum for the arithmetic processing of canceling the speaker sound; and a storage process (S11) for storing the calculated optimum coefficient in the storage means. <P>COPYRIGHT: (C)2010,JPO&INPIT

Description

  The present invention relates to a calibration method for an intercom device.

  2. Description of the Related Art Conventionally, there is an interphone device that performs two-way communication, and includes a speaker that outputs sound from an interphone device installed in another place, a microphone that inputs sound transmitted to another interphone device, and the like.

  Since howling occurs when the sound generated from the speaker wraps around the microphone, various measures for preventing howling are taken. For example, a speaker and a pair of microphones are provided, a delay circuit that delays the output of a microphone closer to the speaker by a delay time of sound waves corresponding to the difference in distance between the two microphones and the speaker, and both of the sounds from the speaker. A level adjustment amplifier circuit that matches the output level of the microphone and a differential amplifier circuit that uses the outputs of both microphones that have passed through the delay circuit and the level adjustment amplifier circuit as both inputs are provided, and the output of the differential amplifier circuit is transmitted. An intercom device has been proposed that uses a signal to cancel speaker sound from a transmitted signal to prevent howling.

  In this intercom device, after picking up the sound from the speakers with both microphones, delay and level adjustment are performed, and the sound components from the speakers that are input to both microphones are canceled by the differential amplifier circuit. Only the components are removed (cancellation process) to prevent howling (for example, see Patent Document 1).

In such an interphone device having a plurality of microphones, it is necessary to calibrate the sensitivity of each microphone to ensure the canceling effect of the speaker sound. For example, the calibration is periodically performed in a field actually used. A method (for example, refer to Patent Document 2) and a method for performing calibration by switching a plurality of ranges in a manufacturing process (for example, refer to Patent Document 3) have been proposed.
Japanese Patent No. 3226121 JP 2004-343700 A JP 2006-135551 A

  However, the method of periodically calibrating in the field actually used as in the above-mentioned Patent Document 2 is not suitable for equipment such as an interphone device, and a plurality of processes in the manufacturing process as in the above-mentioned Patent Document 3. In the method of performing calibration by switching the range, a large measuring device is required, and the work time is long.

  Thus, there is a need for a method for easily calibrating each microphone for an interphone device that prevents howling by canceling speaker sound from a transmission signal using a plurality of microphones.

  The present invention has been made in view of the above reasons, and its purpose is to easily perform microphone calibration of an intercom device that prevents howling by canceling speaker sound from a transmission signal using a plurality of microphones. An object of the present invention is to provide a calibration method for an intercom device that can be performed.

  According to the first aspect of the present invention, there is provided a sounding body that emits sound based on transmitted sound information, first and second microphones that collect sound and output a sound signal, and first and second microphones and sounding body. Calculation processing using a coefficient calculated from a delay time, which is a transmission time of a sound wave corresponding to a difference in distance from each other, and an amplitude ratio of each sound signal of the first and second microphones to sound from the sounding body. A signal processing unit that generates an audio signal in which the sound emitted by the sounding body is canceled by applying it to the audio signals output from the first and second microphones, and a coefficient used by the signal processing unit for arithmetic processing A calibration method for an intercom device that calibrates an intercom device including a storage unit, the intercom device having a sound generator, first and second microphones, and a signal A driving step of driving the sounding body at a specific frequency in a state where the processing unit and the storage unit are incorporated, and the amplitude of each sound signal of the first and second microphones with respect to the sound from the sounding body driven at the specific frequency Based on the amplitude ratio data deriving step for deriving the amplitude ratio data, and the delay based on the phases of the audio signals of the first and second microphones with respect to the sound from the sounding body driven at a specific frequency. A delay time data deriving step for deriving time data; an optimum coefficient calculating step for calculating a coefficient used by the signal processing unit for the arithmetic processing based on each of the derived delay time and the amplitude ratio data; and A storage step of storing the calculated coefficient in the storage unit.

According to this 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 generated by the sounding body incorporated in the interphone device is used. Since the coefficient used for the calculation process for canceling the sound emitted by the sounding body is calculated, it is not necessary to prepare a sounding body separately, and the configuration of the calibration apparatus can be simplified. That is, the microphone of the intercom device that prevents howling by canceling the speaker sound from the transmission signal using a plurality of microphones can be easily calibrated. 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 are performed for each specific frequency sequentially selected from the plurality of frequencies. Each data of the amplitude ratio and the delay time is derived.

  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 is increased.

  According to a third aspect of the present invention, in the second aspect, the optimum coefficient calculating step is based on the amplitude ratio and the delay time data derived for each specific frequency sequentially selected from a plurality of frequencies. A first coefficient used for the calculation process is calculated for each specific frequency, and a plurality of first coefficients calculated for the specific frequency are averaged to calculate a second coefficient used for the calculation process. The storing step stores the second coefficient in the storage unit.

  According to this invention, since only the second coefficient obtained by averaging the first coefficient is calculated, the time required for calibration can be shortened.

  According to a fourth aspect of the present invention, in the third aspect, if the second coefficient is within a predetermined range, the storing step stores the second coefficient in the storage unit, and the second coefficient is within the predetermined range. If it is outside, the second coefficient is not stored in the storage unit.

  According to the present invention, since the second coefficient that is an inappropriate value is not stored in the storage unit, it is possible to ensure the reliability of the intercom apparatus.

  According to a fifth aspect of the present invention, in any one of the first to fourth aspects, 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 the sampling time of the A / D conversion means in advance, and the optimum coefficient calculation step performs the arithmetic processing based on the sampling time read from the storage unit in addition to the delay time and the amplitude ratio data. The coefficient used for is calculated.

  According to the present invention, calibration can be performed for a plurality of types of intercom apparatuses (for example, a plurality of models having different distances between the first and second microphones), and versatility is enhanced.

  As described above, according to the present invention, there is an effect that the microphone can be easily calibrated in the interphone apparatus that prevents howling by canceling the speaker sound from the transmission signal using a plurality of microphones.

  Hereinafter, embodiments of the present invention will be described with reference to the drawings.

(Embodiment 1)
The interphone apparatus A of the present embodiment is shown in FIGS. 13 to 15, and includes an apparatus main body A1 that forms the outline of the apparatus, a speaker module A2 that is attached to the apparatus main body A1, a sound processing module A3, a microphone M1 (first Microphone), microphone M2 (second microphone), and various button switches SW1 to SW3, which are attached to the front face of a box 90 embedded and disposed at an appropriate place in the building. Then, the information line Ls wired through the box 90 is connected, and functions as an intercom device capable of two-way communication between the rooms via the information line Ls. The power supply of the intercom apparatus A may be supplied from an outlet provided near the installation location or may be supplied via the information line Ls. In FIG. 14, some of the constituent elements of the speaker SP described later (such as the yoke 20 and the permanent magnet 22) are omitted.

  The apparatus main body A1 is formed in a substantially box shape with the rear surface and upper and lower surfaces opened by molding ABS (Acrylonitrile-Butadiene-Styrene resin) or PC-ABS (PolyCarbonate-Acrylonitrile-Butadiene-Styreneresin). 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 is further not shown through a notch 102 formed in the approximate center of the upper and lower ends. The apparatus main body A1 is attached to the box 90 by screwing the attachment screw into a screw hole (not shown) of the box 90. And the apparatus main body A1 is provided with attachment holes 103 at the upper and lower ends thereof, and is fixed to a structure such as a wall surface by inserting screws through the attachment holes 103.

  Further, the rear surface of the box-shaped speaker module A2 is locked by the locking pieces 104 and 105 that extend rearward from the upper side of the side surface of the apparatus main body A1 and have locking claws on the inner side, whereby the speaker module A2 is A plurality of sound holes 7 for allowing sound from the speaker SP included in the speaker module A2 to pass through are provided on the front surface of the apparatus main body A1 attached to the inside of the apparatus main body A1 and facing the speaker module A2. In the sound hole 7, a plurality of sound holes 7 are formed in a substantially circular shape in the approximate center of the upper front surface of the apparatus main body A1, and a plurality of recesses 7a are formed around the sound holes by dimple processing.

  Further, the rear surface of the box-shaped sound processing module A3 is locked by a locking piece 106 that extends rearward from the lower side of the side surface of the apparatus main body A1 and has a locking claw on the inner side. It is attached inside the apparatus main body A1. Openings 110, 111, and 112 are provided on the front surface of the apparatus main body A1 facing the voice processing module A3, and protrusions formed on the back surfaces of the call button SW1, the alarm stop button SW2, and the indoor call button SW3, respectively. A pair of attachment holes 103 formed in the lower portion of the apparatus main body A1 are covered with the call button SW1 through the openings 110, 111, 112 from the front and attached to the front surface of the voice processing module A3.

  Further, a plate A4 formed with a plurality of recesses 7a is attached to the upper part of the apparatus main body A1 by locking from the front, and a pair of attachment holes 103 formed in the upper part of the apparatus main body A1 are covered with the plate A4.

  Further, as shown in FIGS. 16A to 16C (the Z1-Z1 cross section of FIG. 16B is shown in FIG. 16A, and the Z2-Z2 cross section of FIG. 16B is shown). 16 (c)), a first cylindrical boss 71 projects from the inner surface of the apparatus main body A1 facing the speaker module A2 substantially at the center of the sound hole 7 formed in a substantially circular shape. A second cylindrical boss 72 projects from the side of the sound hole 7 (where it does not face the sound hole 7), and is formed in a circular recess 71a formed on the end surface of the boss 71 in the axial direction. A microphone M1 is attached, and a microphone M2 is attached to a circular recess 72a formed on an end surface of the boss 72 in the axial direction, so that the microphones M1 and M2 can be easily positioned. Here, the microphones M1 and M2 have a disk-shaped outline, and a rubber cap G is provided on the outer peripheral surface thereof. When the microphones M1 and M2 are press-fitted into the recesses 71a and 72a, the recesses 71a are caused by the elasticity of the rubber cap G. , 72a, the assembly is facilitated without using a fixing means such as an adhesive. Alternatively, if an elastic elastomer is formed on the inner surfaces of the recesses 71a and 72a, it is not necessary to provide the rubber caps G on the microphones M1 and M2, and the microphones M1 and M2 can be easily fixed in the recesses 71a and 72a. it can. Furthermore, the bosses 71 and 72 as a whole may be formed of an elastic elastomer.

  Further, a pair of wirings W1 and W2 for outputting audio signals are led out from the microphones M1 and M2, respectively, and the wirings W1 and W2 are disposed in grooves 71b and 72b formed downward from the recesses 71a and 72a. Thus, it is drawn into the audio processing module M3, preventing interference with the speaker module A2 and securing a wiring path. The microphones M1 and M2 are back electret type electret condenser microphones.

  Thus, the speaker module A2, the sound processing module A3, and the 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 with commercial power supplied from the outside. The AC is converted to an operating power source for an internal circuit composed of a stable DC voltage and supplied to each unit.

  The speaker module A2 is attached to the inner surface of the apparatus main body A1 so as to face the sound hole 7, and as shown in FIG. 14, a resin-molded body A21 having an opening on the rear surface and the opening of the body A21 are covered. A module main body A20 having a width of 40 mm, a height of 30 mm, and a thickness of 8 mm is constituted by the cover A22 formed by resin molding in a flat shape, and a speaker SP is provided in the module main body A20.

  As shown in FIG. 13, the speaker SP has a thickness of about 0.8 mm made of cold rolled steel plate (SPCC, SPCEN), electromagnetic soft iron (SUY) or the like formed on the bottom surface of the circular recess 11 provided on the front surface of the body A21. An annular yoke 20 made 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. .

  A cylindrical permanent magnet 22 (for example, residual magnetic flux density of 1.39 T to 1.43 T) formed of NdFeB is disposed in the opening 20 a in the ring of the yoke 20, and the outer peripheral edge of the dome-shaped diaphragm 23. The part is bonded to the step surface 21 a of the support 21.

  The diaphragm 23 is formed of a thermoplastic plastic (for example, a thickness of 12 μm to 50 μm) such as PET (PolyEthyleneTerephthalate) or PEI (Polyetherimide). A cylindrical bobbin 24 is fixed to the rear surface of the diaphragm 23, and is formed by winding a polyurethane copper wire (for example, φ0.05 mm) around a paper tube of kraft paper at the rear end of the bobbin 24. A voice coil 25 is provided. The bobbin 24 and the voice coil 25 are provided with a cylindrical permanent magnet 22 on the inner side and are provided to face the yoke 20, and freely move in the front-rear direction in the vicinity of the yoke 20.

  Further, as shown in FIG. 14, a terminal plate 30 is disposed on the side of the speaker SP on the front surface of the body A 21, and a pair of terminal portions provided on the terminal plate 30 are connected to a pair of voice coils 25 of the speaker SP. Lead wires are connected by soldering, and output wiring from the audio processing module A3 is similarly connected by soldering.

  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 due to 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. To vibrate in the front-rear direction. At this time, a sound corresponding to the audio signal is emitted from the diaphragm 23. That is, an electrodynamic speaker SP is configured.

  When the speaker SP is attached to the module main body A20, a rear air chamber Br which is a space surrounded by the rear inner side and the inner side surface of the module main body A20 and the rear surface side (yoke 20 side) of the speaker SP is formed. The rear air chamber Br is a space that is insulated from the outside of the module main body A20 by closely contacting the vibration plate 23 of the speaker SP and the stepped surface 21a of the support 21 and further closely contacting the body A21 of the module main body A20 and the cover A22. become.

  When the speaker module A2 is attached to the apparatus main body A1, as shown in FIG. 13, it is surrounded by the inner surface of the apparatus main body A1 and the surface side of the speaker SP (diaphragm 23 side), and is insulated from the rear chamber Br. The microphone M1 is formed in the front air chamber Bf, and the microphone M1 attached to the boss 71 on the back surface of the apparatus main body A1 is disposed in the front air chamber Bf with the sound collection surface facing the substantial center of the diaphragm 23. High directivity with respect to the sound from the speaker SP.

  Further, the boss 72 on the back surface of the apparatus main body A1 fits into the recess 31 provided on the front surface of the body A21 of the speaker module A2, and the boss 72 fits into the recess 31 so that the speaker module A2 is assembled to the apparatus main body A1. Therefore, the positioning can be easily performed, and the assembly work can be performed smoothly. Further, since the microphone M2 attached to the boss 72 communicates with the outside through the sound collection hole 8 drilled in the front surface of the apparatus main body A1, the microphone M2 is transmitted through the sound collection hole 8. And has high directivity with respect to the voice from the speaker located in front of the intercom device A.

  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. The audio signal transmitted from the intercom apparatus A installed in the communication line via the information line Ls is received by the communication unit 81, amplified by the amplification unit 84 via the audio switch unit 82, and then output from the speaker SP. Is done. Further, by operating the call button SW1, a call can be made, and each audio signal input from the microphone M1 and the microphone M2 is subjected to signal processing to be described later in the signal processing unit 85, and then passes through the audio switch unit 83. The data is transmitted from the communication unit 81 to the intercom device A installed in another room or the like via the information line Ls. The alarm stop button SW2 is operated to stop the alarm signal received from the other terminal device via the information line Ls, and the indoor call button SW3 is used to call the interphone device A installed in another room. To operate.

  If the 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 sound output of the speaker SP is the microphone. In order to prevent howling caused by picking up by M1 and M2, the following configuration is provided.

  First, as shown in FIG. 18, the signal processing unit 85 accommodated in the audio processing module A3 includes A / D conversion circuits 85a and 85b for A / D converting the audio signals of the microphones M1 and M2, and the A / D The delay circuit 85c that delays the output of the D conversion circuit 85a, the amplitude adjustment circuit 85d that adjusts the amplitude of the output of the delay circuit 85c, and the amplification circuit 85d that outputs the sound signal of the microphone M2 that is output from the A / D conversion circuit 85b. A subtracting circuit 85e for subtracting the sound signal of the microphone M1.

  19 to 22 show audio signal waveforms of the respective units of the signal processing unit 85 when the audio from the speaker SP is collected by the microphones M1 and M2. First, if the 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. Accordingly, 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 the output 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 output of the microphone M2 is small, and only the sound wave delay time [Td = (X2-X1) / Cv] (Cv is the speed of sound) corresponding to the difference (X2-X1) in the distance between the microphones M1, M2 and the speaker SP. The phase is delayed.

  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. (See FIGS. 19A and 19B).

  The delay circuit 85c is composed of a time delay element or a CR phase delay circuit. By delaying the output of the microphone M1 closer to the speaker SP by the delay time Td, the outputs Y12 and A of the delay circuit 85c are delayed. The phase with the output Y21 of the / D conversion circuit 85b is matched (see FIGS. 20A and 20B).

  Then, the amplitude adjustment circuit 85d generates an output Y13 in which the amplitude of the output Y12 is adjusted. At this time, the amplitude adjustment is performed, and the amplitudes of the output Y13 and the output Y21 with respect to the sound from the speaker SP are matched. The output levels of the two microphones M1 and M2 with respect to the sound from the speaker SP are matched (see FIGS. 21A and 21B). 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 1.

  Then, 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 delay processing and the amplitude adjustment processing, and the subtracting 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). That is, in the output Ya, the sound component from the speaker SP is reduced.

  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 in which the sound collection surface is arranged toward the speaker H. On the other hand, the amplitude of the output Y21 of the microphone M2 is larger than the amplitude of the output Y11 of the microphone M1 with respect to the voice uttered by the speaker H in front of the microphones M1 and 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 is 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 even if the subtraction circuit 85e performs the subtraction process, the output Ya Therefore, the signal corresponding to the voice uttered by the speaker H remains in a state where the amplitude is maintained sufficiently.

  As described above, the audio component from the speaker SP is reduced at the output Ya of the subtracting circuit 85e, and the audio component uttered by the speaker H in front of the interphone device A remains, and the speaker H to be retained is output at the output Ya. The relative difference between the sound component from the speaker component and the sound component from the speaker SP to be reduced increases. That is, even when the sound from the speaker H and the sound from the speaker SP are generated at the same time, only the sound component from the speaker SP is reduced while maintaining a sufficient amplitude for the sound component from the speaker H. Therefore, howling that occurs when the microphones M1 and M2 pick up the sound output of the speaker SP can be prevented.

  However, the delay time Td with respect to the sound from the speaker SP and the amplitudes of the sound signals Y11 and Y21 are ideal under conditions (for example, there is no variation in the point sound source, the housing sealed structure, the CR of the circuit configuration, etc.). It does not depend on the frequency of the sound emitted by the SP, but is constant with respect to the frequency. However, since it is not actually an ideal condition, it depends on the frequency of the sound emitted by the speaker SP.

  Therefore, since the delay time Td between the audio signals Y11 and Y21 with respect to the sound from the speaker SP and the amplitudes 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 represented by an algorithm using a plurality of circuit blocks 85i (i = 1, 2,..., N) corresponding to audio signals having different frequencies and an averaging processing unit 850, as shown in FIG. Can be considered.

  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 R (m ), Each circuit block 85i includes an amplifying 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 adder 93i that adds the outputs of the amplifiers 90i and 91i, an amplifier 94i that amplifies the output of the adder 93i, and the mth digital data R (m ) From the amplifying unit 94i. Then, the averaging processing unit 850 adds the outputs of the subtraction unit 95i of each circuit block 85i and divides by the number n of the circuit blocks 85i, thereby generating an output Ya that averages the outputs of the circuit blocks 85i. .

  The coefficients of 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 having the frequency corresponding to the circuit block 85i. Is set to a value that can. That is, the speaker sound is canceled for each output of the circuit block 85i for each different frequency.

  The output Ya generated by the averaging processing unit 850 is expressed using a total of 3n coefficients (first coefficients) of Bi1, Bi2, and Ci as in [Equation 1].

  Furthermore, since [Equation 1] can be summarized as [Equation 2], the output Ya is obtained by averaging two coefficients (first and second) K1, K2 obtained by averaging Bi1, Bi2, and Ci (first coefficient). 2).

  Therefore, from [Equation 2], the calculation process in the delay circuit 85c and the amplitude adjustment circuit 85d is as shown in FIG. 24. The amplification unit 96 amplifies the m-th digital data S (m) of the microphone M1 with the amplification factor K1. And an amplifying unit 98 that amplifies the m−1th digital data S (m−1) of the microphone M1 through the delay element 97 at an amplification factor K2, and an adding unit 99 that adds the outputs of the amplifying units 96 and 98. The subtracting circuit 85e subtracts the output of the adding unit 99 from the mth digital data R (m) of the microphone M2 to generate the output Ya. In this algorithm, it is only necessary to set two coefficients, that is, the amplification factor K1 of the amplification unit 96 and the amplification factor K2 of the amplification unit 98, and the signal processing algorithm can be simplified as compared with FIG.

  In the present 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, K2] are stored in the storage unit 86 such as a ROM, and the delay circuit 85c and the amplitude The adjustment circuit 85d performs arithmetic processing 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.

  Next, in the voice switch units 82 and 83 (see FIG. 17), the following processing is performed to further prevent howling. The voice switch unit 82 is arranged on the transmission line of the received signal, the voice switch unit 83 is arranged on the transmission line of the signal to be transmitted, and the voice switch units 82 and 83 compare the levels of the input signals with each other. The voice switch section with the smaller input signal level increases the transmission loss on the transmission line by the variable loss means provided inside. Accordingly, a signal having a lower level of the received signal and the transmitted signal is attenuated and the howling margin is further increased, so that further howling prevention is achieved.

  Next, coefficients [amplification factors K1, 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, K2] are stored in the storage unit 86. The calibration stored in will be described.

  As shown in FIG. 2, the calibration apparatus B is installed in an anechoic chamber 200 installed before the packaging process of the production line of the intercom apparatus A. The anechoic chamber 200 has a width of 1.5 ×. It is a rectangular box shape with a depth of 1.5 × height of 2.0 (m), and openings 201 for penetrating the conveyor CV are provided on both side surfaces thereof as shown in FIG. The inside of the anechoic chamber 200 is suitable for calibration because external noise, noise, and the like are blocked.

  As shown in FIG. 4, the inside of the anechoic chamber 200 is covered with a sound absorbing material 202, and the personal computer 211 and the interphone device A being calibrated are fixed on a work desk 210 installed in the vicinity of the conveyor CV. A fixing jig 212 is arranged, and the operator 290 sets the interphone device A before calibration from the conveyor CV to the fixing jig 212.

  As shown in FIG. 5, the personal computer 211 is installed with application software 211a as a measurement analysis tool (hereinafter referred to as measurement analysis software 211a), an A / D conversion unit 211b, a memory writer 211c, a RAM, and the like. The storage unit 211d and the like are provided to constitute measurement calculation means and coefficient writing means.

  Then, the operator 290 operates the personal computer 211 to activate the measurement analysis software 211a, and then performs calibration according to the flowchart of FIG. First, the worker 290 connects the output port 211e of the personal computer 211 to the speaker SP (the voice coil 25) of the interphone 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. In addition, a connector (not shown) may be provided in the intercom apparatus A, and each wiring from the personal computer 211 or the amplifier 220 may be connected to the connector to simplify the connection work.

  Next, the measurement analysis software 211a that has started calibration by the operation of the operator 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 driving process for A / D conversion is started. 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.

  In addition, the storage unit 86 of the interphone device A stores sampling time data of the A / D conversion circuits 85a and 85b in advance, and the measurement analysis software 211a receives the sampling time from the storage unit 86 of the interphone device A at the time of activation. And is set as the sampling time Ts of A / D conversion performed by the A / D conversion unit 211b. That is, the accuracy of calibration is improved by matching the sampling time in the intercom apparatus A with the sampling in the personal computer 211.

  First, the speaker SP is driven by a sine wave of 500 Hz (S2), the speaker sound collected by the microphones M1 and M2 at that time is A / D converted by the A / D converter 211b, and the measurement analysis software 211a is A / D converted. The D-converted signals are stored in the storage unit 211d as the audio signal Ys1 of the microphone M1 and the audio signal Ys2 of the microphone M2 (S3).

  Next, the speaker SP is driven by a sine wave of 1 KHz (S4), and the speaker sound collected by the microphones M1 and M2 at that time is A / D converted by the A / D converter 211b. The signals after the / D conversion are stored in the storage unit 211d as the audio signal Ys1 of the microphone M1 and the audio signal Ys2 of the microphone M2 (S5).

  Next, the speaker SP is driven with a sine wave of 2 KHz (S6), and the speaker sound collected by the microphones M1, M2 at that time is A / D converted by the A / D converter 211b, and the measurement analysis software 211a is A The signals after the / D conversion are stored in the storage unit 211d as the audio signal Ys1 of the microphone M1 and the audio signal Ys2 of the microphone M2 (S7).

  Then, the measurement analysis software 211a starts the amplitude ratio data deriving step and the delay time data deriving step based on the audio signals Ys1 and Ys2 obtained by collecting the 500 Hz, 1 KHz, and 2 KHz sine waves.

  This amplitude ratio data deriving step (S8) will be described below. First, the circuit block 851 in the pre-simplification algorithm shown in FIG. 23 corresponds to a speaker sound of 500 Hz, the circuit block 852 corresponds to a speaker sound of 1 KHz, and the circuit block 853 corresponds to a speaker sound of 2 KHz. Think.

  Then, the audio signal Ys1 output from the microphone M1 that collects the speaker sound of 500 Hz has a peak value V11 (peak to peak) as shown in FIG. 6A, and the microphone M2 that collects the speaker sound of 500 Hz collects the sound signal Ys1. The audio signal Ys2 to be output has a peak value V21 (peak to peak) as shown in FIG. 6B, and the amplitude ratio data at this time is [V11 / V21], and the amplitude ratio data [V11 / V21] is , Equal to the amplification factor C1 of the amplification unit 941. That is, it is expressed by C1 = V11 / V21.

  The audio signal Ys1 output from the microphone M1 that has collected the speaker sound of 1 KHz has a peak value V12 (peak to peak) as shown in FIG. 7A, and is output from the microphone M2 that has collected the speaker sound of 1 KHz. The audio signal Ys2 has a peak value V22 (peak to peak) as shown in FIG. 7B, and the amplitude ratio data at this time is [V12 / V22], and the amplitude ratio data [V12 / V22] is amplified. It becomes equal to the amplification factor C2 of the part 942. That is, it is expressed by C2 = V12 / V22.

  The audio signal Ys1 output by the microphone M1 that has collected the speaker sound of 2 KHz has a peak value V13 (peak to peak) as shown in FIG. 8A, and is output by the microphone M2 that has collected the speaker sound of 2 KHz. 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, it is expressed by C3 = V13 / V23.

  Next, the delay time Td in which the phase of the output of the microphone M2 is delayed with respect to the output of the microphone M1 from the audio signals Y11 and Y21 of the microphones M1 and M2 that collect the speaker sounds of 500 Hz, 1 KHz, and 2 KHz. Derived for each frequency (S9).

  First, as shown in FIGS. 9A and 9B, the audio signals Ys1 and Ys2 output from the microphones M1 and M2 with respect to the speaker sound of 500 Hz are the audio signals Ys2 of the microphone M1 and the audio signals Ys1 of the microphone M1. Is delayed by a delay time Td1.

  As shown in FIGS. 10A and 10B, the audio signals Ys1 and Ys2 output from the microphones M1 and M2 with respect to the speaker sound of 1 KHz are compared with the audio signal Ys1 of the microphone M1 as compared with the audio signal Ys1 of the microphone M1. Is delayed by the delay time Td2.

  As shown in FIGS. 11A and 11B, the audio signals Ys1 and Ys2 output from the microphones M1 and M2 with respect to the speaker sound of 2 KHz are compared with the audio signal Ys1 of the microphone M1 as compared with the audio signal Ys1 of the microphone M1. Is delayed by the delay time Td3.

  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] Then, an optimum coefficient deriving step for deriving optimum coefficients [amplification factors K1, K2] for calculation processing for canceling the speaker sound is started (S10). Here, the following description will be made assuming that the delay times Td1, Td2, and Td3 at 500 Hz, 1 KHz, and 2 KHz have a relationship of Td1 <Td2 <Td3.

  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 set to “1”, and the amplification factor B12 of the amplification unit 921 is set to “0”.

  The amplification factors Bi1 and Bi2 of the circuit block 85i of the circuit block corresponding to another frequency are obtained from 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 with the sampling time Ts, and the plot P1 indicates the mth digital data S output from the microphone M1 corresponding to the frequency with the shortest delay time. (M) is shown, and the plot P2 shows the (m−1) th digital data S (m−1) of the output of the microphone M1 corresponding to the frequency with the shortest delay time. If the difference ΔTd between the shortest delay time and the delay time at another frequency is taken as a difference ΔTd, the plot P3 delayed by the difference ΔTd from the plot P1 on the straight line connecting the plots P1 and P2 corresponds to the other frequency. It is considered that there is m-th digital data output from the microphone M1.

  Thus, [Expression 3] is derived from FIG. 12, and the amplification factors Bi1 and Bi2 are expressed as [Expression 4].

  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 expressed by [Equation 5].

  Further, the amplification factor B31 of the amplification unit 903 and the amplification factor B32 of the amplification unit 923 of the circuit block 853 corresponding to the frequency 2 KHz are expressed by [Equation 6].

  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. Is substituted into [Equation 7] and averaged, and the coefficients [amplification factors K1, K2 used when the delay circuit 85c and the amplitude adjustment circuit 85d of the signal processing unit 85 perform arithmetic processing (the algorithm of FIG. 24). ] Is calculated (S10).

  In this way, the coefficients [amplification factors K1, K2] are calculated in consideration of the frequency characteristics of the sound signals of the microphones M1, M2, so that the calibration accuracy is increased. Further, since only two coefficients (second coefficient) K1 and K2 obtained by averaging Bi1, Bi2, and Ci (first coefficient) are calculated, the time required for calibration can be shortened.

  Here, the measurement analysis software 211a reads sampling time data from the storage unit 86 of the intercom apparatus A at the time of activation, and sets it as the sampling time Ts of A / D conversion performed by the A / D conversion unit 211b. Therefore, the calibration can be performed for a plurality of types of interphone apparatuses A (for example, a plurality of models having different distances between the microphones M1 and M2), and versatility is improved.

  Next, the measurement analysis software 211a of the personal computer 211 determines whether or not the coefficients [amplification factors K1, K2] calculated in S10 are within a preset range, and the coefficients [amplification factors K1, K2] are determined. If it is within the range, the calculated coefficients [amplification factors K1, K2] are stored as appropriate values in the storage unit 86 of the intercom apparatus A by the memory writer 211c. If the coefficient [amplification factor K1, K2] is not within the range, the calculated coefficient [amplification factor K1, K2] is discarded as an inappropriate value, and the inappropriate factor [amplification factor K1, K2] is discarded. Is stored in the storage unit 86, and the reliability of the intercom apparatus A is ensured (S11).

  Then, the operator 290 in the anechoic chamber 200 calculates the coefficients [amplification factors K1, K2] with appropriate values and stores the interphone device A stored in the storage unit 86 on the conveyor CV again. Unload from 200. The work of placing the intercom device A on the conveyor CV may be performed by a robot machine (not shown) to reduce the number of steps. On the other hand, the intercom apparatus A ′ whose coefficient [amplification factor K1, K2] is calculated as an inappropriate value and the calibration is defective is stored in the tray 230 in the anechoic chamber 200 (see FIG. 4). Then, the microphones M1 and M2 are checked for the presence or absence of wiring, the presence or absence of wiring disconnection, etc., and the parts causing the calibration failure are replaced or repaired, and then calibration is performed again.

  The intercom apparatus A that has been calibrated is transported from the anechoic chamber 200 to the packing process by the conveyor C, packed by the operator 300 in the cardboard box 301, and shipped (see FIG. 2).

  As described above, the interphone apparatus A is calibrated in a state where the speaker SP, the microphones M1 and M2, and the sound processing module are assembled to the apparatus main body A1, and uses the sound emitted by the speaker SP incorporated in the interphone apparatus A. Thus, since the coefficients [amplification factors K1, K2] used for the calculation process for canceling the speaker sound are calculated, it is not necessary to prepare a separate sound generator such as a speaker, and the configuration of the calibration apparatus can be simplified.

  In this embodiment, audio signals are exchanged between the intercom apparatuses A using a wired communication system via the information line Ls. However, by providing a well-known wireless communication means in the interphone apparatus A, wireless communication is possible. Voice signals may be exchanged using a communication method.

(Embodiment 2)
In the first embodiment, the external computer 211 and the amplifier 220 in which the measurement analysis software 211a is installed are connected to the intercom apparatus A for calibration, but the intercom apparatus A of the present embodiment is as shown in FIG. In addition, a microcomputer 240 (hereinafter referred to as a microcomputer 240) is stored therein. 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 includes a storage unit 240b such as a RAM. Communication is possible with a terminal 250 such as a personal computer via an external interface 260. The voice processing module A3 includes the functions of the amplifier 220, the A / D conversion unit 211b, and the memory writer 211c of the first embodiment, and the microcomputer 240 and the voice processing module A3 constitute measurement calculation means and coefficient writing means. .

  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 according to the flowchart of FIG. First, the measurement analysis software 240a drives the speaker SP and performs processing in the order of test mode 1 (S21) and test mode 2 (S22).

  First, in the test mode 1, the measurement analysis software 240a validates the input from the microphone M1, invalidates the input from the microphone M2, and then turns the speaker SP in the order of 500 Hz sine wave, 1 KHz sine wave, and 2 KHz sine wave. The sound processing module A3 sequentially performs A / D conversion on the speaker sound collected by the microphone M1 at each frequency, and the measurement analysis software 240a converts the signal at each frequency after the A / D conversion into the sound signal Ys1 of the microphone M1. Is stored in the storage unit 240b.

  Next, in the test mode 2, the measurement analysis software 240a invalidates the input from the microphone M1 and validates the input from the microphone M2, and then the speaker SP in the order of 500 Hz sine wave, 1 KHz sine wave, and 2 KHz sine wave. The sound processing module A3 sequentially performs A / D conversion on the speaker sound collected by the microphone M2 at each frequency, and the measurement analysis software 240a converts each frequency signal after the A / D conversion into the sound signal of the microphone M2. It is stored in the storage unit 240b as Ys2.

  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] (Refer to FIGS. 6 to 11), an optimum coefficient deriving step for deriving optimum coefficients [amplification factors K1, K2] for the calculation process for canceling the speaker sound is performed (S25). The amplitude ratio data deriving step (S23), delay time data deriving step (S24), and optimum coefficient deriving step (S25) are the amplitude ratio data deriving step (S8) and delay time data deriving step (S9) of the first embodiment. ) And the optimum coefficient deriving step (S10), and detailed description thereof is omitted. However, in this embodiment, since the audio signals of the microphones M1 and M2 are individually collected in the test modes 1 and 2, the delay time in which the phase of the output of the microphone M2 is delayed with respect to the output of the microphone M1. Td is derived based on the timing at which the sound of each frequency is output from the speaker SP.

  Next, the measurement analysis software 240a determines whether or not the coefficients [amplification factors K1, K2] calculated in S25 are within a preset range, and the coefficients [amplification factors K1, K2] are 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 apparatus A as an appropriate value. If the coefficient [amplification factor K1, K2] is not within the range, the calculated coefficient [amplification factor K1, K2] is discarded as an inappropriate value, and the inappropriate factor [amplification factor K1, K2] is discarded. Is stored in the storage unit 86, and the reliability of the intercom apparatus A is ensured (S26).

  As described above, in the calibration method of the present embodiment, the intercom apparatus A includes the microcomputer 240 in which the measurement analysis software 240a is installed, and the audio processing module A3 having the functions of the amplifier, the A / D conversion unit, and the memory writer. Therefore, it is not necessary to install measurement analysis software in the external personal computer 211 or prepare the amplifier 220, and the configuration of the calibration apparatus can be further simplified. Furthermore, even after the interphone device A is installed, the calibration according to the installation status can be easily performed only by connecting the terminal device 250 capable of communicating with the microcomputer 240 to the interphone device A. Other configurations are the same as those in the first embodiment, and a description thereof will be omitted.

It is a figure which shows the flowchart of the calibration method for intercom apparatuses of Embodiment 1. FIG. It is a figure which shows the outline | summary of a calibration equipment same as the above. It is a figure which shows the side of an anechoic chamber same as the above. It is the schematic which shows the outline | summary in 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 waveform figure of a 500-Hz audio | voice signal same as the above. (A) (b) It is a wave form diagram of a 1-kHz audio | voice signal same as the above. (A) (b) It is a wave form diagram of a 2kHz audio | voice signal same as the above. (A) (b) It is a waveform figure of a 500-Hz audio | voice signal same as the above. (A) (b) It is a wave form diagram of a 1-kHz audio | voice signal same as the above. (A) (b) It is a wave form diagram of a 2kHz 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 an 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 the reverse view which expanded a part of apparatus main body same as the above. It is a block diagram of a voice processing module same as the above. It is a circuit block 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. (A) (b) It is a signal waveform diagram of a signal processing part same as the above. It is a signal waveform figure of a signal processing part same as the above. It is a figure which shows the algorithm before simplification in a delay circuit and an amplitude adjustment circuit same as the above. It is a figure which shows the algorithm after the simplification in the delay circuit and amplitude adjustment circuit same as the above. It is a figure which shows the structure of the calibration apparatus of Embodiment 2. FIG. It is a figure which shows the structure of a microcomputer same as the above. It is a figure which shows the flowchart of the calibration method for interphone apparatuses same as the above.

Explanation of symbols

A Intercom device SP Speaker M1, M2 Microphone 211 Personal computer 220 Amplifier

Claims (5)

A sounding body that emits sound based on the transmitted sound information;
First and second microphones for collecting sound and outputting sound signals;
From the delay time which is the transmission time of the sound wave corresponding to the difference between the distances between the first and second microphones and the sounding body, and the amplitude ratio of the sound signals of the first and second microphones to the sound from the sounding body. A signal processing unit that generates an audio signal in which the sound emitted by the sounding body is canceled by performing arithmetic processing using the calculated coefficient on the audio signal output from the first and second microphones;
A storage unit that stores coefficients used by the signal processing unit for arithmetic processing;
A calibration method for an intercom device that calibrates an intercom device equipped with
A driving step of driving the sounding body at a specific frequency in a state in which the sounding body, the first and second microphones, the signal processing unit, and the storage unit are incorporated in the intercom device;
An amplitude ratio data deriving step for deriving the data of the amplitude ratio based on the amplitude of each audio signal of the first and second microphones with respect to the sound from the sounding body driven at a specific frequency;
A delay time data deriving step for deriving the delay time data based on the phase of each sound signal of the first and second microphones with respect to the sound from the sounding body driven at a specific frequency;
An optimum coefficient calculating step for calculating a coefficient used by the signal processing unit for the arithmetic processing based on each of the derived delay time and amplitude ratio data;
A storage step of storing the calculated coefficient in a storage unit;
A calibration method for an intercom device, comprising: The driving step drives the sounding body at a specific frequency sequentially selected from a plurality of frequencies,
The amplitude ratio data deriving step and the delay time data deriving step derive each data of the amplitude ratio and the delay time for each specific frequency sequentially selected from a plurality of frequencies. Calibration method for intercom equipment. The optimal coefficient calculating step uses a first data used for the calculation processing for each specific frequency based on the amplitude ratio and the delay time data derived for each specific frequency sequentially selected from a plurality of frequencies. And calculating a second coefficient used for the arithmetic processing by averaging the plurality of first coefficients calculated for each specific frequency,
The intercom device calibration method according to claim 2, wherein the storing step stores the second coefficient in the storage unit.   The storage step stores the second coefficient in the storage unit if the second coefficient is within a predetermined range, and stores the second coefficient if the second coefficient is outside the predetermined range. 4. The intercom device calibration method according to claim 3, wherein the interphone device calibration method is not stored in the unit. 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,
5. The optimum coefficient calculating step calculates a coefficient used for the arithmetic processing based on a sampling time read from a storage unit in addition to each data of the delay time and the amplitude ratio. The interphone apparatus calibration method according to any one of the above.

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