JP2010074541A - Intercom device - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide an intercom device easily improving cancel processing performance of speaker sound even when a transmission function of sound waves transmitted from a speaker to a microphone has a frequency characteristic, and preventing howling. <P>SOLUTION: A signal processing part 85 includes: A/D conversion circuits 85c, 85c for making phases of respective audio signals of microphones M1, M2 coincide with each other in a predetermined frequency by delaying A/D conversion timing of the audio signal of the microphone M2; an amplitude adjustment means using an adaptive filter 91 for making the respective output levels of the microphones M1, M2 for speaker sound coincide with each other; an arithmetic part 92 outputting n pieces of audio signals being differences between the respective audio signals of the microphones M1, M2 having passed through the A/D conversion circuits 85c, 85c and the adaptive filter 91; and a signal selection part 93 selecting and outputting a minimum signal from the n pieces of audio signals. <P>COPYRIGHT: (C)2010,JPO&INPIT

Description

  The present invention relates to an intercom apparatus.

  2. Description of the Related Art Conventionally, there is an interphone device installed indoors, which 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 was proposed as a signal.

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).
Japanese Patent No. 3226121

  However, the sound emitted from the speaker is transmitted to the microphone and converted into an audio signal, but the transfer function of the sound wave transmitted from the speaker to the microphone has a frequency characteristic, and the phase of the sound collected by the microphone, The amplitude depends on the frequency. This has been a cause of deteriorating speaker sound canceling performance.

  The present invention has been made in view of the above reasons, and the purpose thereof is to easily improve the cancellation performance of speaker sound even if the transfer function of the sound wave transmitted from the speaker to the microphone has frequency characteristics. An object of the present invention is to provide an intercom device capable of preventing howling.

  According to the first aspect of the present invention, the speaker that outputs the transmitted sound information, the first microphone that collects the sound and outputs the sound signal, and the distance from the speaker are arranged at a position farther than the first microphone. A first microphone that collects voice and outputs a voice signal; and a signal processing unit that processes and transmits each voice signal output from the first and second microphones, A transmission time of a sound wave having a predetermined frequency corresponding to a difference between each distance between the second microphone and the speaker is set as a delay time, and the signal processing unit performs first and second microphones on the sound from the speaker according to the delay time. Delay time adjusting means for matching the phases of the respective audio signals at the predetermined frequency, and an adaptive frame for adjusting the amplitude ratio of the respective audio signals of the first and second microphones to the audio from the speakers. An amplitude adjusting means using a filter, a delay time adjusting means, and a computing means for outputting the difference between the audio signals of the first and second microphones that have passed through the amplitude adjusting means. While changing the amplitude ratio of the audio signals of the first and second microphones with respect to the audio, the arithmetic unit calculates the difference between the audio signals of the first and second microphones. A sequential optimum adjustment is performed in which the amplitude ratio of each audio signal is set to an amplitude ratio that minimizes the calculation result.

  According to the present invention, by using the adaptive filter in the amplitude adjusting means for matching the output levels of the first and second microphones with the sound from the speaker, the transfer function of the sound wave transmitted from the speaker to the microphone has frequency characteristics. Even if it does, the cancellation process performance of a speaker sound can improve easily and howling can be prevented.

  According to a second aspect of the present invention, in the first aspect, an amplitude ratio obtained by dividing the sound signal of the first microphone with respect to the sound from the speaker by the sound signal of the second microphone is equal to or greater than a first predetermined value, and the amplitude The speaker and the first and second microphones are arranged so that a value obtained by dividing a ratio fluctuation range with respect to a frequency by a center value of the fluctuation range is equal to or less than a second predetermined value. .

  According to the present invention, the deterioration of the speaker sound included in the sound signal output from the signal processing unit is suppressed, and the S / N ratio (the sound component of only the speaker sound from the front is S, The noise component generated by successively adjusting the amplitude ratio of each output of the second microphone to N can be maintained.

  According to a third aspect of the present invention, in the first or second aspect, the delay time adjusting means includes a first A / D conversion means for A / D converting an audio signal output from the first microphone, and a second microphone. Second A / D conversion means for A / D converting the audio signal output from the first microphone, and the delay from the timing when the first A / D conversion means A / D converts the audio signal from the first microphone. When time elapses, the second A / D conversion means performs A / D conversion on the audio signal from the second microphone.

  According to the present invention, even when an A / D conversion unit having a low sampling frequency is used as the first and second A / D conversion units, the delay time can be accurately adjusted.

  According to a fourth aspect of the present invention, in any one of the first to third aspects, the signal processing unit changes the phase difference between the audio signals of the first and second microphones while the first and second operations are performed by the arithmetic unit. And a phase adjustment means for performing sequential optimum adjustment for calculating a difference between the sound signals of the first and second microphones and setting a phase difference between the sound signals of the first and second microphones to a phase difference that minimizes the calculation result. It is characterized by that.

  According to this invention, not only the frequency response of the amplitude of the audio signal collected by the first and second microphones but also the frequency response of the delay time is taken into account, so that the amplitude correction and delay time correction of the audio signal are performed. Further, the effect of canceling the speaker sound is further improved, and the howling prevention effect is further improved.

  According to a fifth aspect of the present invention, in the fourth aspect, the first A / D conversion means for A / D converting the audio signal output from the first microphone, and the audio signal output from the second microphone as the A / D. A second A / D conversion means for converting, and the phase adjustment means includes two digital signals output continuously from at least one of the first and second A / D conversion means. It is determined that the audio signal is continuously changing along a straight line connecting the values, and the first and second microphones are changed by the arithmetic unit while changing the phase difference between the audio signals of the first and second microphones. When calculating the difference between the audio signals, the value on the straight line is used as the audio signal of at least one of the A / D conversion means.

  According to the present invention, since the amplitude of the audio signal is estimated by linear approximation, phase adjustment can be easily performed with a simple configuration.

  A sixth aspect of the present invention is the signal processing device according to any one of the first to fifth aspects, wherein the signal processing unit is configured to perform at least the delay time adjustment unit, the amplitude adjustment unit, and the calculation when a signal input to the speaker is equal to or less than a predetermined threshold The canceling process of the speaker sound by the means is not performed.

  According to the present invention, if it is determined that the level of the speaker sound is equal to or lower than the predetermined threshold, the speaker sound canceling process is stopped, and the deterioration of the speaker sound due to the unnecessary canceling process can be prevented.

  As described above, in the present invention, even if the transfer function of the sound wave transmitted from the speaker to the microphone has a frequency characteristic, the effect of canceling the speaker sound can be easily improved and howling can be prevented. There is.

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

(Embodiment 1)
The interphone device J of the present embodiment is shown in FIGS. 8 to 10, and includes a device main body J1 that forms an outline of the device, a speaker module J2, a sound processing module J3, and a microphone M1 (first microphone) attached to the device main body J1. Microphone), microphone M2 (second microphone), and various button switches SW1 to SW3, which are attached to the front surface of a box BX embedded and disposed at an appropriate place in the building. And the information line Ls wired via the box BX is connected, and it has a bidirectional | two-way telephone call function via the information line Ls between rooms. The power supply of the intercom device J may be supplied from an outlet provided in the vicinity of the installation place or may be supplied via the information line Ls. In FIG. 9, 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 J1 is formed in a substantially box shape with the rear surface and the 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 BX, and is further not shown via a notch 102 formed at the approximate center of the upper and lower ends. The apparatus main body J1 is attached to the box BX by screwing the attachment screw into a screw hole (not shown) of the box BX. And the apparatus main body J1 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 J2 is locked by the locking pieces 104 and 105 extending rearward from the upper side of the side surface of the apparatus main body J1 and having locking claws on the inner side, whereby the speaker module J2 is A plurality of sound holes 7 for allowing sound from the speaker SP included in the speaker module J2 to pass through are provided on the front surface of the apparatus main body J1 attached to the inside of the apparatus main body J1 and facing the speaker module J2. 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 J1, and a plurality of recesses 7a are formed around the sound hole 7 by dimple processing.

  Further, the rear surface of the box-like sound processing module J3 is locked by a locking piece 106 that extends rearward from the lower side of the side surface of the apparatus main body J1 and has a locking claw on the inner side thereof. It is attached inside the apparatus body J1. Openings 110, 111, and 112 are provided on the front surface of the apparatus main body J1 facing the voice processing module J3, 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 part of the apparatus main body J1 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 J3.

  Further, a plate J4 formed with a plurality of recesses 7a is attached to the upper part of the apparatus main body J1 by locking from the front, and a pair of attachment holes 103 formed in the upper part of the apparatus main body J1 are covered with the plate J4.

  Further, as shown in FIGS. 11A to 11C (note that the Z1-Z1 cross section of FIG. 11B is shown in FIG. 11A, and the Z2-Z2 cross section of FIG. 11 (c)), a first cylindrical boss 71 projects from the inner surface of the apparatus main body J1 facing the speaker module J2 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 the end surface of the boss 72 in the axial direction (see FIG. 8), 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. Then, it is drawn into the audio processing module J3, preventing interference with the speaker module J2, and securing a wiring path. The microphones M1 and M2 are back electret type electret condenser microphones.

  Thus, the speaker module J2, the sound processing module J3, and the microphones M1 and M2 are attached to the inner surface of the apparatus main body J1, and a power supply circuit (not shown) provided in the box BX is supplied with commercial power supplied from the outside. Is converted to an operating power supply for an internal circuit composed of a stable DC voltage and supplied to each part.

  The speaker module J2 is attached to the inner surface of the apparatus main body J1 so as to face the sound hole 7, and as shown in FIGS. 12 to 14, a body J21 formed of resin with an opening provided on the rear surface, and the body J21 A module body J20 having a width of 40 mm, a height of 30 mm, and a thickness of 8 mm is constituted by a flat resin-molded cover J22 covering the opening, and a speaker SP is provided in the module body J20.

  As shown in FIG. 8, 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 J21. An annular yoke 20 made of an iron-based material and a cylindrical support body 21 extending forward from the outer peripheral edge of the yoke 20 are integrally formed in the recess 11 of the body J21. .

  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.

  A voice signal is input to the voice coil 25 via a pair of lead wires W3 (see FIG. 12), and the lead wire W is connected to the voice coil 25 at one end side of the circular diaphragm 23 on the back side. The other end side is led out of the module body J20 through the space between the diaphragm 23 and the stepped surface 21a of the support body 21 in the radial direction.

  12 to 14, on the front surface of the body J21, a terminal plate 30 is disposed on the bottom surface of the recess 12 provided on the side of the recess 11 in which the speaker SP is disposed. The other ends of the pair of lead wires W3 are connected to the pair of terminal portions 30a and 30b by soldering, and the output wiring from the sound processing module J3 is also connected to the pair of terminal portions 30a and 30b by soldering. Therefore, the process of connecting the lead wire W3 to the output wiring from the sound processing module J3 and the maintenance at the time of disconnection can be easily performed.

  When an audio signal is input to the polyurethane copper wire of the voice coil 25 via the lead wire W3, 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 The diaphragm 23 is vibrated 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 J20, a rear air chamber Br that is a space surrounded by the rear inner side and the inner side surface of the module main body J20 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 body J20 by closely contacting the diaphragm 23 of the speaker SP and the stepped surface 21a of the support body 21 and further contacting the body J21 and the cover J22 of the module body J20. become.

  Furthermore, on the inner surface of the module main body J20, as shown in FIG. 14A, one end is separated from the inner wall surface along the inner wall surface of the body J21 surrounding the rear air chamber Br on the back surface of the speaker SP, and the other end is A wall portion 41 that is continuous with the inner wall surface is erected, and the end surface 51 of the wall portion 41 abuts against the back surface of the cover J22, so that the wall portion 41, the inner wall surface of the body J21, and the back surface of the cover J22 Thus, a hollow acoustic tube 40 is formed, and this acoustic tube 40 is arranged in a small-capacity rear air chamber Br. The acoustic tube 40 is a hollow closed tube having a rectangular cross-sectional shape bent along the inner wall surface of the rear air chamber Br, and is formed by opening one end (open end 40a) and closing the other end (closed end 40b). The inside of the pipe communicates with the rear air chamber Br through the open end 40a. An acoustic tube uses the fact that the input impedance becomes extremely small at the resonance frequency of the closed tube (the frequency at which the total length of the tube coincides with an odd multiple of a quarter wavelength). The reflected wave has a waveform whose phase is inverted with respect to the incident wave, and the incident wave and the reflected wave cancel each other, thereby reducing the sound wave propagating from the opening end 40a to the outside.

  Such an acoustic tube 40 is provided to shift the lowest resonance frequency of the speaker SP to the low frequency side and further increase the sound pressure level of the speaker SP. Is set to approximately a quarter wavelength of a low frequency (in the present embodiment, around 700 to 800 Hz), the sound quality and efficiency of the speaker SP can be improved even if the rear air chamber Br has a small capacity.

  In addition, ribs 52 are protruded from the four corners of the joint surface 50 constituting the outer periphery of the rear surface of the body J21 (see FIGS. 14A and 14B), and the cover J22 covers the rear opening of the body J21. The chamfered four corners 62 (see FIG. 13) of the cover J22 abut on the inside of the rib 52.

  When this speaker module J2 is attached to the apparatus main body J1, as shown in FIG. 8, it is surrounded by the inner surface of the apparatus main body J1 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 J1 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 J1 is fitted into a recess 31 provided on the front surface of the body J21 of the speaker module J2, and the boss 72 is fitted into the recess 31 so that the speaker module J2 is assembled to the apparatus main body J1. Therefore, the positioning can be easily performed, and the assembly work can be performed smoothly. Furthermore, 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 J1, the microphone M2 is transmitted through the sound collection hole 8. The directivity is high with respect to the voice from the speaker located in front of the intercom device J.

  Next, as shown in FIG. 15, the audio processing module J3 is composed of an IC including a communication unit 81, audio switch units 82 and 83, an amplification unit 84, and a signal processing unit 85, and is installed in another room or the like. The audio signal transmitted from the intercom apparatus J 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. 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 J 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 J 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.

  As shown in FIG. 1, the signal processing unit 85 accommodated in the audio processing module J3 includes amplification circuits 85a and 85b, A / D conversion circuits 85c and 85d, a timing control unit 85e, and an audio processing circuit 85f. Consists of. Each operation of the A / D conversion circuits 85c and 85d is controlled by the timing control unit 85e, and the operation of each circuit will be described below.

  First, the amplifier circuits 85a and 85b amplify each audio signal output from the microphones M1 and M2 with a predetermined amplification factor, the audio signal Y11 (t) from the amplified microphone M1, and the audio signal from the amplified microphone M2. Y21 (t) is output.

  Here, the microphone M1 has a sound collection surface mounted toward the speaker SP, and the speaker SP is more effective than the sound (speaker sound) emitted by the speaker H (see FIG. 8) located in front of the interphone device J. Collects sound emitted by with high sensitivity. On the other hand, the microphone M2 has a sound collection surface arranged forward, and collects sound emitted by the speaker H located in front of the interphone device J with higher sensitivity than the sound emitted by the speaker SP.

  That is, the sound signal Y11 (t) from the microphone M1 has high sensitivity and large amplitude for the sound emitted from the speaker SP, but has low sensitivity and small amplitude for the sound emitted from the speaker H. Further, the sound signal Y21 (t) from the microphone M2 has high sensitivity and large amplitude with respect to the sound emitted from the speaker H, but has low sensitivity and small amplitude with respect to the sound emitted from the speaker SP.

  Furthermore, at the time of transmission, both the sound emitted by the speaker H and the sound emitted by the speaker SP are collected by the microphones M1 and M2, but the distance X1 from the center of the speaker SP to the center of each of the microphones M1 and M2 , X2 is X1 <X2, so that for the sound from the speaker SP, a phase difference occurs between the sound signal Y11 (t) from the microphone M1 and the sound signal Y21 (t) from the microphone M2. The sound signal Y11 of the microphone M1 (Vs is the speed of sound) corresponding to the sound wave delay time [Td = (X2-X1) / Vs] (Vs is the speed of sound) corresponding to the distance difference (X2-X1) between the microphones M1, M2 and the speaker SP Compared with t), the phase of the audio signal Y21 (t) of the microphone M2 is delayed (see FIGS. 2A and 2B). The delay time Td is constant with respect to the frequency without depending on the frequency of the sound emitted from the speaker SP under ideal conditions (for example, a point sound source, a sealed housing structure, and no variation in CR of the circuit configuration). Although there is actually no ideal condition, it depends on the frequency of the sound emitted by the speaker SP.

  Similarly, the amplitudes of the audio signals Y11 (t) and Y21 (t) with respect to the sound from the speaker SP are constant with respect to the frequency under the ideal conditions without depending on the frequency of the sound emitted from the speaker SP. Although there is actually no ideal condition, it depends on the frequency of the sound emitted by the speaker SP.

  On the other hand, since the distances between the microphones M1 and M2 and the speaker H can be regarded as being equal to each other, the voice signals Y11 (t) and Y21 (t) of both microphones M1 and M2 are obtained for the voice emitted by the speaker H. The phase is substantially the same.

  The A / D conversion circuit 85c converts the amplified analog audio signal Y11 (t) of the microphone M1 into a digital signal, and the A / D conversion circuit 85d converts the analog audio signal Y21 of the amplified microphone M2. (T) is converted into a digital signal. The A / D conversion operation of the A / D conversion circuit 85c is performed in synchronization with the rise of the control signal Si1 output from the timing control unit 85e, and the A / D conversion operation of the A / D conversion circuit 85d is performed by the timing control unit. This is performed in synchronization with the rise of the control signal Si2 output by 85e. Furthermore, the phase of the audio signal Y21 (t) of the microphone M2 is delayed by a delay time Td1 relative to the audio of the frequency f1 (reference frequency) emitted by the speaker SP as compared to the audio signal Y11 (t) of the microphone M1. Therefore, the operation of the timing controller 85e is set in advance so that the clock signal Si2 is generated with a delay time Td1 with respect to the clock signal Si1, so that the audio signals Y11 (t ), Y21 (t) is A / D converted at the same phase at the frequency f1 (FIGS. 2C and 2D).

  Therefore, even with the low sampling frequency A / D conversion circuits 85c and 85d, it is possible to accurately adjust the delay time, and the audio signals Y11 (t) and Y21 (t) of the microphones M1 and M2 at the frequency f1. The phases can be matched with high accuracy.

  The A / D conversion circuit 85c outputs an audio signal Y12 (t) composed of digital values a1, b1, c1, d1, e1,... Obtained by A / D converting the audio signal of the microphone M1. (See FIG. 2 (e)). From the A / D conversion circuit 85d, an audio signal composed of digital values a2, b2, c2, d2, e2,... Obtained by A / D converting the audio signal of the microphone M2. Y22 (t) is output (see FIG. 2 (f)), the digital value a1 of the microphone M1, the digital value a2 of the microphone M2, the digital value b1 of the microphone M1, the digital value b2 of the microphone M2, and the digital value of the microphone M1. By making the value c1 correspond to the digital value c2,... Of the microphone M2, respectively, the audio signals of the microphones M1, M2 have the same phase at the frequency f1. There.

However, as described above, the amplitudes of the audio signals Y11 (t) and Y21 (t) with respect to the sound from the speaker SP depend on the frequency of the sound emitted from the speaker SP, and the sound emitted from the speaker SP is the microphone. If the transfer function transmitted to M1 is Hs1 (ω), and the transfer function transmitted from the speaker SP to the microphone M2 is Hs2 (ω), the amplitude ratio A of the audio signals Y11 (t) and Y21 (t) ( ω) (hereinafter referred to as audio signal amplitude ratio A (ω)) is
A (ω) = | Hs1 (ω) | / | Hs2 (ω) | (1)
It is expressed.

FIG. 3 shows the frequency characteristic of the audio signal amplitude ratio A (ω) of the present embodiment, and the audio signal amplitude ratio A (ω) is within an amplitude ratio fluctuation range ΔL between 500 and 3500 (Hz). It has fluctuated. The amplitude ratio fluctuation width ΔL is set such that the maximum value of the audio signal amplitude ratio A (ω) is max [A (ω)] and the minimum value is min [A (ω)].
ΔL = max [A (ω)] − min [A (ω)] (2)
It is expressed.

The sound processing circuit 85f of the present embodiment includes an amplitude adjusting unit using an adaptive filter that matches the output levels of the microphones M1 and M2 with respect to the sound from the speaker SP, and a specific algorithm thereof is as follows. The range of min [A (ω)] to max [A (ω)] is divided into n stages (eg, divided into n stages at equal intervals), and the reciprocal of the audio signal amplitude ratio [A ( ω)] ^{− 1} is prepared as the amplitude correction coefficient Bi (where i = 1, 2,..., n), and the adaptive filter unit 91 of the amplitude sequential optimum adjustment type receives the audio signal Y12 from the microphone M1 ( An audio signal Y13i (t) (where i = 1, 2,..., n) is generated by multiplying each amplitude correction coefficient Bi by t). The adaptive filter unit 91 includes a delay time element 91a and an amplification unit 91b.

The amplitude correction coefficient Bi is
B ^ i = [min [A (ω)] + {ΔL / n} (i−1)] ⁻¹ (3)
(Where i = 1, 2,..., N)
It is represented by Here, if the delay time Td is constant regardless of the frequency, the amplitude correction error is within the range of the value of {ΔL / n} in the second term on the right side of the above equation (3), and this value should be reduced. Thus, the canceling ability of the speaker sound can be improved.

  Next, the computing unit 92 subtracts the sound signal Y22 (t) of the microphone M2 from the sound signal Y13i (t) of the microphone M1, thereby canceling the sound signal Yai (t) ( However, i = 1, 2,..., N) is generated.

  The signal selector 93 selects the minimum signal from the n audio signals Yai (t), and outputs the selected minimum audio signal Yai (t) as the output signal Yo (t). That is, the output signal Yo (t) is output from the speaker SP by the adaptive filter unit 91 adjusting the amplitude of the audio signal of the microphone M1 using the optimum amplitude correction coefficient Bi among the n amplitude correction coefficients Bi. The audio signal has the most canceled audio component.

  On the other hand, for the sound uttered by the speaker H in front of the microphones M1 and M2, the amplitude of the sound signal Y21 of the microphone M2 having the sound collection surface arranged toward the speaker H is such that the sound collection surface faces the speaker SP. It becomes larger than the amplitude of the audio signal Y11 of the arranged microphone M1. Further, since the signal from the microphone M1 is attenuated by the adaptive filter unit 91, the voice component from the speaker H included in the voice signal Y22 (t) is the voice from the speaker H included in the voice signal Y13i (t). Even larger than the ingredients. That is, the amplitude difference between the speech component from the speaker H included in the speech signal Y22 (t) and the speech component from the speaker H included in the speech signal Y13i (t) becomes large, and the subtracting unit 92 performs the subtraction. Even if the processing is performed, a signal corresponding to the voice uttered by the speaker H remains in the voice signal Yai (t) while maintaining a sufficient amplitude.

  In the audio signal Yo output from the signal processing unit 85 as described above, the audio component from the speaker SP is reduced, while the audio component emitted from the speaker H in front of the interphone device J toward the microphones M1 and M2 remains. In the audio signal Yo, the relative difference between the audio component from the speaker H to be retained and the audio component from the speaker SP to be reduced increases over a wide frequency band, and the audio output of the speaker SP is Howling that occurs when M1 and M2 pick up can be prevented.

  Furthermore, by using the adaptive filter unit 91 as the amplitude adjusting means for matching the output levels of the microphones M1 and M2 with the sound from the speaker SP, the transfer function of the sound wave transmitted from the speaker SP to the microphones M1 and M2 has a frequency characteristic. Even if it has, the cancellation processing performance of the speaker sound can be improved easily, and an excellent howling prevention effect can be obtained.

  Next, conditions for making the howling prevention effect by the above algorithm for canceling the speaker sound by the sound processing circuit 85f more effective in the present embodiment will be described.

First, assuming that speech (speaker sound) uttered from the speaker H in front of the interphone device J toward the microphones M1 and M2 is input to the microphones M1 and M2 at the same timing and at the same level, the signal processing unit 85, Y11 (t) = Y21 (t) = Ys (t), and the audio signal Yo (t) output from the signal processing unit 85 when only the speaker sound is input to the microphones M1 and M2 is obtained. ,
Yo (t) = Ys (t) −Ys (t) · exp (−jω (Td1)) · (Ao ⁻¹ + Rm)
= Ys (t) {1−Ao− ¹ · exp (−jω (Td1)) − Rm · exp (−jω (Td1))} (4)
It becomes.

Here, Ao is the center value of the fluctuation range with respect to the frequency of the audio signal amplitude ratio A (ω) (hereinafter referred to as the fluctuation range center value Ao).
Ao = {max [A (ω)] + min [A (ω)]} / 2 (5)
(However, Ao> 1)
It is represented by

Rm is an effective value of noise caused by a fluctuation range with respect to the frequency of the audio signal amplitude ratio A (ω) when the effective value of the speaker sound is 1, (hereinafter referred to as the effective noise value Rm).
Rm = ΔL / Ao (6)
It is represented by

In the above equation (4), [Ao ⁻¹ · exp (−jω (Td1))] is a deterioration factor of speaker sound, and [Rm · exp (−jω (Td1))] is an increase in noise component. It can be seen as minutes.

As a worst condition, when the delay time Td1 = 0 corrected by the A / D conversion timing of the A / D conversion circuits 85c and 85d, the speech component S of only the speaker sound from the front that is originally desired, and the microphone M1, Each signal level of the noise component N generated by sequentially optimally adjusting the amplitude ratio of each output of M2 is:
S = 20 · Log (1-Ao ⁻¹ )
N = 20 · Log (Rm) ……… (7)
It becomes. Therefore, in this embodiment, the S / N ratio after the signal processing by the above algorithm of the audio processing circuit 85f is
S / N = 20 · Log {(1-Ao ⁻¹ ) / (Rm)} (8)
It becomes.

  FIG. 4 shows the relationship between the audio signal amplitude ratio A (ω) and the speaker sound deterioration amount. For example, in order to reduce the speaker sound deterioration amount to 3 dB or less, the audio signal amplitude ratio A (ω) Is required to be 3.42 or more.

  FIG. 5 shows the relationship between the amplitude ratio fluctuation width ΔL and the fluctuation width center value Ao when min [A (ω)] = 3.42 is fixed. FIG. 6 shows Rm (= ΔL / The relationship between Ao) and the S / N ratio is shown. For example, in order to secure the S / N ratio of 6 dB or more, it is necessary to maintain Rm of 0.38 or less.

  Thus, in order to obtain the desired speaker sound deterioration amount and speaker sound S / N ratio from the above equations (4) to (6), the frequency of the sound signal amplitude ratio A (ω) is determined. The center value of the fluctuation range: Ao, and the effective value of the noise: Rm generated by the fluctuation width with respect to the frequency of the audio signal amplitude ratio A (ω) may be set as shown in FIG. That is, the speaker SP and the microphones M1 and M2 may be arranged so as to satisfy the respective values of Ao and Rm (both relative position and absolute position).

  Although specific numerical values of Ao and Rm are shown in FIG. 7, the speaker sound canceling effect by this algorithm increases by increasing Ao and decreasing Rm. In other words, when speech (speaker sound) uttered from the speaker H in front of the interphone device J toward the microphones M1 and M2 is input to the microphones M1 and M2 at the same timing and at the same level. If Ao is large and Rm is small, not only the speaker sound but also the speaker sound is greatly canceled, and the noise level is greatly increased, that is, each condition as shown in FIG. 7 is satisfied. By setting Ao and Rm to, the present algorithm becomes effective.

  Next, the voice switch sections 82 and 83 (see FIG. 15) further prevent howling by performing the following processing. The voice switch unit 82 is arranged on the transmission line of the received signal, and the voice switch unit 83 is arranged on the transmission line of the signal to be transmitted to constitute the voice switch processing unit 86, and the voice switch units 82 and 83. Compare the levels of the input signals with each other, and the voice switch section with the smaller input signal level increases the transmission loss on the transmission line by the variable loss means provided therein. 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.

  4 to 7, voices (speaker sounds) emitted from the speaker H in front of the interphone device J toward the microphones M1 and M2 are input to the microphones M1 and M2 at the same timing and at the same level. However, the present invention can obtain the same effect regardless of the presence or absence of this precondition.

(Embodiment 2)
In the first embodiment, the delay time Td, which is the time difference from when the sound emitted from the speaker SP reaches the microphone M1 until it reaches the microphone M2, is not constant with respect to the frequency as shown in FIG. When the delay time with respect to the frequency is Td (ω), the transfer function that the sound emitted from the speaker SP is transmitted to the microphone M1 is Hs1 (ω), and the transfer function that the sound emitted from the speaker SP is transmitted to the microphone M2 is Hs2 (ω).
Td (ω) = (2π / ω) · {tan ⁻¹ [Hs2 (ω)] − tan ⁻¹ [Hs1 (ω)]} / 360 (9)
It is represented by In FIG. 16, the fluctuation range between the maximum value max [Td (ω)] and the minimum value min [Td (ω)] of the delay time Td (ω) for this frequency is represented by ΔTd (hereinafter referred to as “Td”). Delay time variation width ΔTd).

Here, the reference delay time To is
To = {max [Td (ω)] + min [Td (ω)]} / 2 (10)
It is represented by

  Then, the sound processing circuit 85f of the signal processing unit 85 calculates the difference between the sound signals of the microphones M1 and M2 while changing the phase difference of the sound signals of the microphones M1 and M2 around the reference delay time To. 17 includes phase adjustment means for performing sequential optimum adjustment for setting the phase difference between the audio signals M1 and M2 to the phase difference that minimizes the calculation result. The phase adjustment means is specifically shown in FIG. In this manner, the audio processing circuit 85f is configured by adding a phase sequential optimum adjustment type adaptive filter unit 94. The adaptive filter unit 94 is a linear prediction FIR filter, and includes an amplification unit 94a, a delay time element 94b and an amplification unit 94c, a delay time element 94d, and an amplification unit 94e.

  The audio signal Y22 (t) of the microphone M2 is composed of digital values a2, b2, c2, d2, e2,... Obtained by A / D converting analog values, as shown in FIG. The operation of the adaptive filter unit 94 will be described using digital values a2, b2, and c2 shown in FIG.

  First, the adaptive filter unit 94 approximates the amplitude of the audio signal Y22 (t) between the digital values a2-b2 of the microphone M2 with a straight line K1, and the amplitude of the audio signal Y22 (t) between the digital values b2-c2 is a straight line. Approximate with K2. Then, the delay time fluctuation width ΔTd is set to m stages [. . . To-2, To-1, To, To + 1, To + 2,. . . (For example, ΔTd is divided into m steps at equal intervals), and the reference delay time To, which is the center of the m steps, is set to the advance side and the delay side of the digital value b2 in correspondence with the digital value b2. The amplitude of the audio signal in the total m stages of delay time Tj (where j = 1, 2,..., M) is estimated from the linearly approximated straight lines K1, K2, and the audio signal Y23ij of the microphone M2 is estimated. (T) (where i = 1, 2,..., N, j = 1, 2,..., M). At this time, the phase correction coefficients in the amplification unit 94a, amplification unit 94c, and amplification unit 94e are C1ij, C2ij, and C3ij, respectively.

  Then, the calculation unit 92 generates the audio signal Yaij (t) in which the audio component from the speaker SP is canceled by subtracting the audio signal Y23ij (t) of the microphone M2 from the audio signal Y13i (t) of the microphone M1. To do.

  Then, the signal selection unit 93 selects the minimum signal from the n × m audio signals Yaij (t), and outputs the selected minimum audio signal Yaij (t) as the output signal Yo (t). That is, the output signal Yo (t) has an audio component from the speaker SP using a combination of n amplitude correction coefficients Bi and m delay times Tj and using an optimum amplitude correction coefficient Bi and delay time Tj. It becomes the most canceled audio signal.

  Therefore, not only the frequency characteristic of the amplitude of the audio signal collected by the microphones M1 and M2, but also the frequency characteristic of the delay time is taken into account, and the sound signal amplitude correction and delay time correction are performed. This further improves the howling prevention effect. Further, since the amplitude of the audio signal of the microphone M2 is estimated by linear approximation, phase adjustment can be easily performed with a simple configuration.

  In the present embodiment, the number of parameters to be processed by the voice processing circuit 85f is n × m, and it is necessary to perform a lot of arithmetic processing. Therefore, the circuit scale of the voice processing circuit 85f may increase, but n , M, for example, if the calculation is devised using a total of six parameters including the fluctuation width center value Ao, the reference delay time To, the amplitude ratio fluctuation width ΔL, and the delay time fluctuation width ΔTd, a small circuit scale can be obtained. Therefore, it is possible to realize with a low load. For example, when the CPU is not used, the voice processing circuit 85f can be configured with a simple digital circuit, and when the CPU is used, the voice processing circuit 85f can be realized by low-load arithmetic processing.

(Embodiment 3)
FIG. 19 shows a specific configuration of the voice switch processing unit 86 (see FIG. 15) in the first or second embodiment, which is arranged on the transmission line of the signal received by the voice switch unit 82 and transmitted by the voice switch unit 83. The comparator 86a compares the levels of the input signals of the voice switch units 82 and 83, and the voice switch unit with the smaller input signal level is provided inside the variable line. Transmission loss on the transmission line is increased by loss means. 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.

  In the present embodiment, the comparison result of the comparator 86a is further output to the signal processing unit 85. If the signal processing unit 85 determines from the comparison result that the level of the speaker sound is equal to or lower than a predetermined threshold value, the audio processing circuit 85f. The speaker sound canceling process by the (amplitude sequential optimal adjustment type adaptive filter unit 91, arithmetic unit 92, signal selection unit 93, phase sequential optimal adjustment type adaptive filter unit 94) is stopped, and the speaker sound is generated by unnecessary cancellation processing. Prevents deterioration.

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

It is a block diagram of the signal processing part of the intercom apparatus of Embodiment 1. (A)-(f) It is a wave form diagram of each part of a signal processing part same as the above. It is a figure which shows the frequency characteristic of an audio signal amplitude ratio same as the above. It is a figure which shows the relationship between an audio signal amplitude ratio same as the above, and the amount of speaker sound degradation. It is a figure which shows the relationship between an amplitude ratio fluctuation range same as the above, and an amplitude ratio. It is a figure which shows the relationship between a noise effective value same as the above and S / N ratio. It is a figure which shows the setting of each parameter 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 perspective view which shows a speaker module same as the above. It is a disassembled perspective view which shows a speaker module same as the above. (A) (b) It is a top view which shows a body same as the above. It is a block diagram of a voice processing module same as the above. It is a figure which shows the frequency characteristic of the delay time of Embodiment 2. It is a block diagram of a signal processing part same as the above. It is a figure which shows the linear prediction of an audio | voice signal same as the above. It is a figure which shows the structure of the voice switch process part periphery of Embodiment 3. FIG.

Explanation of symbols

J Interphone device SP Speaker M1, M2 Microphone 85 Signal processing unit 85c, 85d A / D conversion circuit 85f Audio processing circuit 91 Adaptive filter 92 Operation unit 93 Signal selection unit

Claims (6)

A speaker that outputs the transmitted audio information, a first microphone that collects audio and outputs an audio signal, and a distance from the speaker that is far from the first microphone and collects audio A second microphone that outputs an audio signal, and a signal processing unit that processes and transmits each audio signal output from the first and second microphones, and the first and second microphones and a speaker, The transmission time of a sound wave of a predetermined frequency corresponding to the difference in each distance is a delay time,
The signal processor
Delay time adjusting means for matching the phases of the audio signals of the first and second microphones with respect to the sound from the speaker at the predetermined frequency according to the delay time;
Amplitude adjusting means using an adaptive filter that adjusts the amplitude ratio of each audio signal of the first and second microphones to the audio from the speaker;
Calculating means for outputting a difference between the audio signals of the first and second microphones that have passed through the delay time adjusting means and the amplitude adjusting means,
The amplitude adjusting means calculates the difference between the audio signals of the first and second microphones by the calculation unit while changing the amplitude ratio of the audio signals of the first and second microphones with respect to the sound from the speaker. An intercom apparatus characterized by performing sequential optimal adjustment in which the amplitude ratio of each audio signal of the first and second microphones is set to an amplitude ratio that minimizes the calculation result.   The amplitude ratio obtained by dividing the sound signal of the first microphone with respect to the sound from the speaker by the sound signal of the second microphone is equal to or greater than a first predetermined value, and the fluctuation range of the amplitude ratio with respect to the frequency is changed. 2. The intercom apparatus according to claim 1, wherein the speaker and the first and second microphones are arranged so that a value obtained by dividing the width by a center value is equal to or less than a second predetermined value.   The delay time adjusting means includes a first A / D converting means for A / D converting an audio signal output from the first microphone, and a second A / D converting the audio signal output from the second microphone. A / D conversion means, and when the delay time elapses from the timing when the first A / D conversion means A / D converts the audio signal from the first microphone, the second A / D conversion means The intercom apparatus according to claim 1 or 2, wherein A / D-converts an audio signal from the second microphone.   The signal processing unit calculates a difference between the audio signals of the first and second microphones by the calculation unit while changing a phase difference between the audio signals of the first and second microphones, The intercom apparatus according to any one of claims 1 to 3, further comprising phase adjustment means for performing sequential optimum adjustment for setting a phase difference between the audio signals of the two microphones to a phase difference that minimizes the calculation result. .   First A / D conversion means for A / D converting the sound signal output from the first microphone, and second A / D conversion means for A / D converting the sound signal output from the second microphone. The phase adjusting means includes a voice signal continuous along a straight line connecting two digital values continuously output by at least one of the first and second A / D converting means. When calculating the difference between the audio signals of the first and second microphones by changing the phase difference of the audio signals of the first and second microphones while changing the phase difference between the audio signals of the first and second microphones. 5. The intercom apparatus according to claim 4, wherein a value on the straight line is used as an audio signal of at least one of the A / D conversion means.   The signal processing unit does not perform speaker sound cancellation processing by at least the delay time adjusting unit, the amplitude adjusting unit, and the calculating unit when a signal input to the speaker is a predetermined threshold value or less. Item 6. An intercom device according to any one of Items 1 to 5.

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