JP4853382B2 - Telephone device - Google Patents

Description

  The present invention relates to a call device used for an interphone or the like.

  Conventionally, there is a communication device installed indoors with an interphone system or the like, which includes a speaker that outputs sound from a communication device installed in another place, a microphone that inputs sound transmitted to the other communication device, and the like. Yes. Then, since howling occurs when the sound generated from the speaker goes around the microphone, the howling is avoided by separating the distance between the speaker and the microphone so that the microphone does not pick up the sound emitted from the speaker. .

  However, there is a problem that if the distance between the speaker and the microphone is increased, the apparatus itself becomes larger and it is difficult to reduce the size. Therefore, in order to achieve both miniaturization of the device and countermeasures for howling, for example, one microphone is arranged at the center of the speaker diaphragm, and the acoustic signal generated from the surface of the speaker diaphragm and the back of the diaphragm are generated. Some acoustic signals cancel each other out in front of the microphone, thereby reducing the substantial sensitivity of the microphone to the sound emitted by the diaphragm of the speaker and preventing howling. (For example, refer to Patent Document 1).

  As another configuration, a pair of microphones, a delay circuit that delays the output of the microphone closer to the speaker by the delay time of the sound wave corresponding to the difference in distance between the two microphones and the speaker, and both the microphone and the speaker Adjusting the level corresponding to the difference in distance to match the output level of both microphones with respect to the sound from the speaker, and using both microphone outputs through the delay circuit and the level adjusting amplifier circuit as both inputs There has also been proposed a communication device that includes a differential amplifier circuit and uses the output of the differential amplifier circuit as a transmission signal.

In this communication device, after picking up the sound from the speaker with a pair of microphones, the delay and level adjustment is performed so that the sound component from the speaker input to both microphones is canceled by the differential amplifier circuit. The transmitted voice can be transmitted in a state in which only the voice component from the speaker is removed and no reception blocking occurs. (For example, refer to Patent Document 2).
JP 2004-320399 A (paragraph number [0037], FIG. 2) Japanese Patent No. 2607257 (page 2, left column, line 13 to right column, third line, page 4, right column, lines 26 to 49, FIGS. 1 and 5)

  However, in the conventional example of Patent Document 1, the phase difference between the acoustic signal generated from the surface of the diaphragm of the speaker and the acoustic signal generated from the back surface of the diaphragm does not become 180 ° at the front of the microphone. It was difficult to cancel each other's acoustic signals in front of each other, and howling could not be prevented completely. Further, since the microphone is incorporated in the speaker, it is difficult to reduce the size and thickness of the speaker, and it is difficult to reduce the size and thickness of the communication device itself.

  Also, in the conventional example of Patent Document 2 described above, in order to prevent howling, the distance between the speaker and the microphone cannot be made very small, and further downsizing of the apparatus has been desired.

  The present invention has been made in view of the above-described reasons, and an object of the present invention is to provide a communication device capable of preventing howling and downsizing.

  According to the first aspect of the present invention, a speaker that outputs sound information transmitted from the outside, first and second microphones that collect sound and output sound signals, and first and second microphones output. The first microphone is a Rayleigh determined by the area of the diaphragm of the speaker and the wavelength of the sound radiated from the diaphragm. The second microphone is disposed within the length, the second microphone is disposed with the sound collection unit facing out of the main body, and the sound processing unit uses the sound signal output from the first microphone to output the sound output from the second microphone. It is characterized in that processing for reducing the sound emitted from the speaker from the signal is performed.

  According to the present invention, the sound wave radiated from the diaphragm of the speaker travels as a plane wave from the diaphragm to the Rayleigh length and becomes substantially constant without attenuating the sound pressure. Since the distance attenuates in inverse proportion to the distance from the sound source, the sound emitted from the speaker can be reliably collected by the first microphone arranged within the Rayleigh length, and the howling prevention processing by the sound processing unit can be reliably performed. it can. Furthermore, since the second microphone directs the sound collection unit out of the housing, the acoustic coupling between the speaker and the second microphone is reduced, and the second microphone becomes difficult to collect the sound emitted by the speaker. The microphone can be arranged in the vicinity of the speaker, and the communication device can be further downsized. That is, howling can be prevented and downsizing can be achieved.

  According to a second aspect of the present invention, in the first aspect, the first microphone is arranged in the main body with a sound collecting portion facing a diaphragm of the speaker.

  According to the present invention, the sound emitted from the speaker can be easily generated even if the sound emitted from the speaker is given directivity by the first microphone disposed in the main body with the sound collecting portion facing the diaphragm of the speaker. Can collect sound.

  According to a third aspect of the present invention, in the first or second aspect, the first microphone faces the center of the diaphragm of the speaker.

  According to the present invention, it is possible to correctly collect the phase of the sound emitted from the speaker.

  As described above, the present invention has the effect of preventing howling and reducing the size.

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

(Embodiment)
The communication device A according to the present embodiment is shown in FIGS. 1 to 4, and includes a housing A1 (horizontal dimension X1 = 40 mm, vertical dimension) including a body A10 having an opening on the front surface and a cover A11 covering the opening of the body A10. X2 = 90 mm, thickness dimension X3 = 8 mm), and an interphone device that includes a speaker SP, a microphone board MB1, a call switch SW1, and a voice processing unit 10 in a housing A1, and enables two-way calls between rooms. It functions. Note that the power of the communication device A may be supplied from an outlet provided in the vicinity of the installation location or may be supplied via the information line Ls.

  As shown in FIG. 4, the audio processing unit 10 is composed of an IC including a communication unit 10a, echo cancellation units 10b and 10c, an amplification unit 10d, and a signal processing unit 10e, and is arranged in the housing A1. A voice signal transmitted from the communication device A installed in another room or the like via the information line Ls is received by the communication unit 10a, amplified by the amplification unit 10d via the echo cancellation unit 10b, and then the speaker. Output from SP. Further, by operating the call switch SW1, a call can be made and each audio signal input from the microphone M1 (first microphone) and the microphone M2 (second microphone) on the microphone board MB1 is received by the signal processing unit 10e. After being subjected to signal processing to be described later, the signal passes through the echo cancel unit 10c, and is transmitted from the communication unit 10a to the communication device A installed in another room or the like via the information line Ls. That is, it functions as an intercom that allows two-way calls between rooms. Note that the power of the communication device A may be supplied from an outlet provided in the vicinity of the installation location or may be supplied via the information line Ls.

  As shown in FIGS. 1 to 3, the speaker SP includes a yoke (iron-based material having a thickness of about 0.8 mm such as cold rolled steel plate (SPCC, SPKEN), electromagnetic soft iron (SUY)) or a permanent magnet (column). A support body 20 provided with a type neodymium magnet, a residual magnetic flux density of 1.39 T to 1.43 T) and the like, and an edge on the outer peripheral side of the dome-shaped diaphragm 21 is fixed to the support body 20. The diaphragm 21 is formed of a thermoplastic plastic (for example, 35 μm to 50 μm in thickness) such as PEN (polyethylene naphthalate) or PEI (polyetherimide), and is a voice coil (craft paper not shown) fixed to the back of the diaphragm 21. When an audio signal is input to a paper tube having a polyurethane copper wire wound around), an electromagnetic force is generated in the voice coil due to the current of the audio signal and the magnetic field of the permanent magnet of the support 20, so that the diaphragm 21 Is vibrated in the front-rear direction. At this time, a sound corresponding to the audio signal is emitted from the diaphragm 21. That is, an electrodynamic speaker SP is formed (for example, a diameter of 25 mm and a thickness of 3.6 mm). Since the configuration of the electrodynamic speaker is well known, detailed description thereof is omitted.

  The mounting holes 22 provided at the four corners of the support 20 of the speaker SP are placed on the four cylindrical bosses 11 formed at equal intervals on the inner surface of the cover A11, and are formed in the axial direction of the bosses 11. By screwing the attachment screw 23 into the screw hole 11a, the speaker SP is fixed in a state where the diaphragm 21 faces the inner surface of the cover A11. In addition, a plurality of sound holes 12 are drilled at locations facing the diaphragm 21 of the cover A11.

  When the cover A11 to which the speaker SP is fixed is attached to the body A10 and the housing A1 is assembled, two spaces A1a and A1b partitioned by the partition wall 13 are formed in the housing A1 by the partition wall 13 standing on the bottom surface of the body A10. And the speaker SP is disposed in the space A1a. The space A1a is formed in a space corresponding to the horizontal dimension X1 = 40 mm, the vertical dimension X2a = 30 mm, and the thickness dimension X3 = 8 mm of the housing A1, and the space A1b is the horizontal dimension X1 = 40 mm and the vertical dimension X2a of the housing A1. = 60 mm and a thickness corresponding to a thickness dimension X3 = 8 mm.

  In the space A1a, the front air chamber Bf that is a space surrounded by the inner surface of the cover A11 and the surface side of the speaker SP (diaphragm 21 side), the bottom surface of the body A10, and four inner wall surfaces (including the partition wall 13). ) And the rear air chamber Br, which is a space surrounded by the back side of the speaker SP (the support 20 side). The front air chamber Bf communicates with the outside through a plurality of sound holes 12 provided in the cover A11. The rear air chamber Br becomes a space that is insulated (not communicated) with the front air chamber Bf by the support 20 of the speaker SP being in close contact with the inner surface of the cover A11, and the cover A11 is formed in the front opening of the body A10. By being in close contact, it is a sealed space that is insulated from the outside.

  Next, as shown in FIG. 5, the microphone substrate MB1 includes a module substrate 2 having both surfaces 2a and 2b. A pair of the microphone bare chip BC1 and ICKa1 and a pair of the microphone bare chip BC2 and ICKa2 are connected to the module substrate 2. And mounted between the bare chips BC1, ICKa1, and the wiring pattern (not shown) on the module substrate 2 and between the bare chips BC2, ICKa2 and the wiring pattern (not shown) on the module substrate 2 respectively. After each connection (wire bonding) with W, the shield case SC1 is mounted so as to cover the pair of bare chips BC1 and ICKa1, and the shield case SC2 is mounted so as to cover the pair of bare chips BC2 and ICKa2. BC1, ICKa1, and shield case SC1 Microphones M1 to be provided with a bare chip BC2, ICKa2, microphone M2 composed of the shield case SC2.

  As shown in FIG. 6, in the bare chip BC (bare chip BC1 or BC2), an Si thin film 1d is formed on one surface side of the silicon substrate 1b so as to close the hole 1c formed in the silicon substrate 1b. An electrode 1f is formed between them via an air gap 1e, and a pad 1g for outputting an audio signal is further provided to constitute a capacitor type silicon microphone. Then, an external acoustic signal vibrates the Si thin film 1d, whereby the capacitance between the Si thin film 1d and the electrode 1f changes to change the amount of charge, and the pad 1g changes with this change in the amount of charge. , 1g, a current corresponding to the acoustic signal flows. In this bare chip BC, the silicon substrate 1b is die-bonded on the module substrate 2. In particular, the Si thin film 1d of the bare chip BC2 faces the sound hole F2 formed in the module substrate 2.

  The microphone M1 uses the bottom surface side of the shield case SC1 with the sound hole F1 as a sound collecting surface, and the microphone M2 uses the mounting surface side with respect to the module substrate 2 with the sound hole F2 as a sound collecting surface. The sound collecting surfaces are provided on both sides of the module substrate 2 in opposite directions. Since the microphone substrate MB1 configured in this manner has both the microphones M1 and M2 mounted on the one surface 2a of the module substrate 2, the thickness of the microphone substrate MB1 can be reduced.

  FIG. 7A is a plan view of the microphone board MB1 as viewed from the one surface 2a side of the module board 2. The module board 2 has a rectangular part 2f in which the microphone M1 is arranged and a rectangular part 2g in which the microphone M2 is arranged. And a connecting portion 2h that connects the rectangular portions 2f and 2g, and the rectangular portion 2g is formed larger than the rectangular portion 2f. A negative power supply pad P1, a positive power supply pad P2, an output 1 pad P3, and an output 2 pad P4 are provided along the edge of the rectangular portion 2g.

  As shown in FIG. 7B, the negative power supply pad P1 is connected to the negative side of the power supply voltage supplied from the outside, and the positive power supply pad P2 is connected to the positive side of the power supply voltage. Power is supplied to the microphones M1 and M2 through the pattern. Further, an audio signal collected by the microphone M1 is output from the output 1 pad P3 via a wiring pattern on the module substrate 2, and an audio signal collected by the microphone M2 is output from the output 2 pad P4. It is output via the upper wiring pattern. The ground of the audio signal output from the output pads P3 and P4 is shared by the negative power supply pad P1.

  Thus, the power of the microphones M1 and M2 is supplied from the common negative power supply pad P1 and the positive power supply pad P2, and the ground of each output of the microphones M1 and M2 is shared by the negative power supply pad P1, so that the number of pads The configuration can be simplified.

  Next, the operation of the microphone substrate MB1 will be described.

  First, each current flowing from the bare chips BC1 and BC2 according to the collected acoustic signal is impedance-converted by ICKa1 and Ka2 and converted into a voltage signal, and output from the output 1 pad P3 and the output 2 pad P4 as audio signals, respectively. Is done.

  The ICKa (ICKa1 or ICKa2) has the circuit configuration of FIG. 8, and is a constant IC composed of a chip IC that converts a power supply voltage + V (for example, 5V) supplied from the power supply pads P1 and P2 into a constant voltage Vr (for example, 12V). A voltage circuit Kb is provided, a constant voltage Vr is applied to the series circuit of the resistor R11 and the bare chip BC, and a connection midpoint between the resistor R11 and the bare chip BC is connected to the junction type J-FET element S11 via the capacitor C11. Connected to the gate terminal. The drain terminal of the J-FET element S11 is connected to the operating power supply + V, and the source terminal is connected to the negative side of the power supply voltage via the resistor R12. Here, the J-FET element S11 is for electrical impedance conversion, and the voltage at the source terminal of the J-FET element S11 is output as an audio signal. Note that the ICKa impedance conversion circuit is not limited to the above-described configuration, and may be, for example, a circuit having a function of a source follower circuit using an operational amplifier, or an audio signal amplification circuit in the ICKa if necessary. May be provided.

  The microphone board MB1 can efficiently configure the signal lines and the power supply lines by performing signal transmission and power supply via the wiring pattern on the module board 2 as described above, and can be attached to the outer surface of the housing A1. Become. In the present embodiment, one surface 2a of the module substrate 2 is arranged along the outer front surface of the housing A1, and the microphone M1 is inserted through the opening 14 on the front surface of the housing A1 so that the sound collection surface faces the front air chamber Bf, and the shield The sound hole F1 of the microphone M1 drilled in the bottom surface of the case SC1 faces the center of the diaphragm 21 of the speaker SP and has high directivity with respect to the sound from the speaker SP transmitted through the sound hole F1. Therefore, the sound emitted from the speaker SP can be reliably collected. Furthermore, by making the sound collection surface of the microphone M1 face the center of the diaphragm 21 of the speaker SP, the microphone M1 can easily collect the sound emitted by the speaker SP in a correct phase.

  The microphone M2 is fitted into a recess 15 provided on the front surface of the housing A1, and the sound hole F2 of the microphone M2 formed in the module substrate 2 is located outside (frontward) the housing A1 in the output direction of the speaker SP. Therefore, it has high directivity with respect to the voice transmitted from the speaker located in front of the communication device A, which is transmitted through the sound hole F2. If the distances from the center of the speaker SP to the centers of the microphones M1 and M2 are X1 and X2, respectively, X1 <X2.

  Further, the rear air chamber Br facing the back surface of the speaker SP is sealed in the housing A1, so that sound radiated from the back surface of the speaker SP is difficult to leak from the rear air chamber Br, and the speaker SP and the microphone M2 are not connected. Acoustic coupling is reduced. Furthermore, the sound radiated from the back surface of the speaker SP (the back surface of the diaphragm 21) has a phase reversed from that of the sound radiated from the surface of the speaker SP (the surface of the diaphragm 21). When the generated sound circulates forward, the sound radiated from the surface of the speaker SP cancels each other, the radiated sound pressure of the speaker SP decreases, and the speaker in front cannot hear the sound emitted by the speaker SP. However, since the sound radiated from the back surface of the speaker SP is difficult to leak to the outside of the housing A1 as described above, a decrease in the radiated sound pressure of the speaker SP due to the wraparound is prevented.

  Further, since the recessed portion 15 in which the microphone M2 is accommodated is a separated space that does not communicate with the rear air chamber Br, the microphone M2 becomes more difficult to collect the sound emitted by the speaker SP, and the speaker SP and the microphone M2 are separated. The acoustic coupling is further reduced. That is, with the above configuration, the sound emitted from the speaker SP and the sound emitted from the speaker are separated and collected by the microphones M1 and M2.

  Further, when the microphone substrate MB1 is disposed in the housing A1, it is difficult to maintain the spatial insulation between the front air chamber Bf and the rear air chamber Br, but the microphone substrate MB1 is disposed in the housing as in the present embodiment. By attaching to the outer surface of A1, the spatial insulation between the front air chamber Bf and the rear air chamber Br can be maintained.

  And in this embodiment, in order to prevent the howling which generate | occur | produces when the microphones M1 and M2 pick up the audio | voice output of the speaker SP, it has the following structures.

  First, as shown in FIG. 9, the signal processing unit 10 e housed in the audio processing unit 10 includes an amplification circuit 30 that non-inverts and amplifies the output of the microphone M <b> 1, and an audio band (300 to 4000 Hz) from the output of the amplification circuit 30. ), A delay circuit 32 that delays the output of the bandpass filter 31, an amplifier circuit 33 that inverts and amplifies the output of the microphone M2, and an audio band from the output of the amplifier circuit 33. A band-pass filter 34 that removes noise at frequencies other than (300 to 4000 Hz), and an adder circuit 35 that adds outputs of the delay circuit 32 and the band-pass filter 34 are provided.

  10 to 13 show the sound signal waveforms of the respective parts of the signal processing unit 10 when the sound from the speakers is collected by the microphones M1 and M2. First, when the distances from the center of the speaker SP to the centers of the microphones M1 and M2 are X1 and X2, respectively, X1 <X2. Therefore, when the sound from the speaker SP is picked up by the microphones M1 and M2, the output Y21 of the microphone M2 is more than the output Y11 of the microphone M1 depending on the distance between the speaker SP and the microphones M1 and M2 and the directivity of the microphones M1 and M2. And the microphone M2 for the delay time [Td = (X2-X1) / Vs] (Vs 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 of the output Y21 is delayed (see FIGS. 10A and 10B).

  The amplifier circuit 30 generates an output Y12 obtained by non-inverting amplification of the output Y11, and the amplifier circuit 33 generates an output Y22 obtained by inverting and amplifying the output Y21 and inverting the phase by 180 °. At this time, level adjustment corresponding to the difference (X2−X1) in the distance between the two microphones M1, M2 and the speaker SP is performed to match the output levels of the two microphones M1, M2 with respect to the sound from the speaker SP (FIG. 11 ( a) (b)). In the present embodiment, the amplification factor of the amplifier circuit 30 is approximately 1, and the amplifier circuit 30 may be omitted.

  Then, the band pass filters 31 and 34 generate outputs Y13 and Y23 obtained by removing noise of frequencies other than the audio band from the outputs Y12 and Y22 (see FIGS. 12A and 12B).

  Next, the delay circuit 32 is composed of a time delay element or a CR phase delay circuit, and delays the output of the microphone M1 closer to the speaker SP by the delay time Td, so that the output Y14 of the delay circuit 32 and The phase with the output Y23 of the bandpass filter 34 is matched to reduce noise on the transmitted audio signal.

  Then, the sound component from the speaker SP included in the output Y14 and the sound component from the speaker SP included in the output Y23 have the same amplitude and the same phase by the amplification process and the delay process. By adding Y23, an output Ya in which the audio signal corresponding to the audio from the speaker SP is canceled is generated (see FIGS. 13A to 13C). That is, in the output Ya, the sound component from the speaker SP is reduced.

  On the other hand, for the sound emitted by the speaker H in front of the microphones M1, M2, the amplitude of the output Y21 of the microphone M2 having high directivity with respect to the sound emitted by the speaker H is larger than the amplitude of the output Y11 of the microphone M1. Also grows. Furthermore, since the amplification factor of the amplification circuit 33 is larger than the amplification factor of the amplification circuit 30, the speech component from the speaker H included in the output Y23 is further larger than the speech component from the speaker H included in the output Y14. That is, the amplitude difference between the speech component from the speaker H included in the output Y14 and the speech component from the speaker H included in the output Y23 becomes large, and even if the addition process is performed by the addition circuit 35, 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 sound component from the speaker SP is reduced at the output Ya of the adder circuit 35, and the sound component emitted from the speaker H in front of the communication device A toward the microphone board MB1 remains, and at the output Ya The relative difference between the speech component from the speaker H to be retained and the speech 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.

  In this way, the sound signal collected by the pair of microphones M1 and M2 is input to the signal processing unit 10e to prevent howling. In particular, the sound collection characteristic of the microphone M1 that picks up the speaker sound is important. Therefore, in the present embodiment, the position of the microphone M1 with respect to the speaker SP is set as follows.

The sound wave radiated forward from the diaphragm 21 of the speaker SP travels as a plane wave from the diaphragm 21 to the Rayleigh length and becomes substantially constant without attenuating the sound pressure. When the Rayleigh length is exceeded, the sound pressure is increased by spherical diffusion. The distance attenuates in inverse proportion to the distance from the sound source. With Rayleigh
Rayleigh length = area of diaphragm 21 (sound generator) / wavelength of radiated sound wave, and the range in which the sound wave radiated from diaphragm 21 maintains a sound pressure above a certain value is within the Rayleigh length. . Thus, in the present embodiment, the microphone M1 is disposed within the Rayleigh length, and the distance Z (see FIG. 1) from the front surface of the diaphragm 21 to the sound collection surface of the microphone M1 is equal to or less than the Rayleigh length with respect to the voice band. It is set to become. Therefore, since the microphone M1 can reliably collect the speaker sound in a plane wave state in which the sound pressure is hardly attenuated, the howling prevention processing by the signal processing unit 10e of the audio processing unit 10 is reliably performed. Even if the sound emitted from the speaker SP has directivity, the speaker sound can be reliably collected by the microphone M1 disposed within the Rayleigh length.

Specifically, if the area of the diaphragm 21 used for the speaker SP having a diameter of 25 mm is 491 mm ² , the frequency band to be used is 600 to 3000 Hz, and the sound velocity is 340 m / sec, the wavelength of the emitted sound wave is 113 to 567 mm. The Rayleigh length is 0.87 to 4.35 mm. Therefore, the distance Z from the front surface of the diaphragm 21 to the sound collection surface of the microphone M1 is set to about 0.9 mm.

  FIG. 14 shows the amount of speaker sound canceled by the signal processing unit 10e. The characteristic 101 is the amount of cancellation in this embodiment in which the microphone M1 is arranged within the Rayleigh length, and the characteristic 102 is the case where the microphone M1 is arranged outside the Rayleigh length. Indicates the amount of cancellation. In the characteristic 101 of the present embodiment, the amount of cancellation is larger than that of the characteristic 102 particularly in the region of 1000 Hz or more, and the effect of arranging the microphone M1 within the Rayleigh length appears. Further, by arranging the microphone M1 within the Rayleigh length, the communication device A can be reduced in size.

  Furthermore, since the microphone M2 has the sound collection surface facing the outside of the housing A1, and the output direction of the speaker SP and the directivity of the microphone M2 are substantially the same direction, the acoustic coupling between the speaker SP and the microphone M2 is reduced, and the microphone M2 Makes it difficult to collect the sound emitted by the speaker SP, and the microphone M2 can be arranged adjacent to the speaker SP, that is, adjacent to the front air chamber Bf, so that the communication device A can be further reduced in size.

  Next, the audio signal output from the adder circuit 35 is output to the echo cancel unit 10c, and the echo cancel units 10b and 10c (see FIG. 4) perform further processing to prevent howling.

  First, the echo canceling unit 10c captures the output of the echo canceling unit 10b as a reference signal, and performs an operation on the output of the signal processing unit 10e, thereby further processing the audio signal that has circulated from the speaker SP to the microphones M1 and M2. Cancel. On the other hand, the echo canceling unit 10b also captures the output of the echo canceling unit 10c as a reference signal and performs an operation on the output of the communication unit 10a, thereby wrapping the audio signal from the speaker to the microphone on the other party side Cancel.

  Specifically, the echo cancellation units 10b and 10c are configured by a speaker SP-microphone M1, M2-signal processing unit 10e-echo cancellation unit 10c-communication unit 10a-echo cancellation unit 10b-amplification unit 10d-speaker SP. By adjusting the amount of loss in a variable loss means (not shown) provided in the loop circuit, the loop gain is set to 1 or less to prevent howling. Here, it is assumed that the smaller signal level of the transmission signal and the reception signal is not important, and the transmission loss of the variable loss circuit inserted in the transmission line having the smaller signal level is increased.

  In the present embodiment, voice signals are exchanged between the call devices A using a wired communication system via the information line Ls. However, by providing the call devices A with known wireless communication means, Voice signals may be exchanged using a communication method.

(A) (b) (c) It is a figure which shows the structure of embodiment. It is a perspective view same as the above. It is a partially exploded view same as the above. It is a circuit block diagram of a communication apparatus same as the above. It is side surface sectional drawing which shows the structure of a microphone substrate same as the above. It is side surface sectional drawing which shows the structure of a bare chip same as the above. It is the (a) simplified top view and (b) simplified circuit diagram which show the structure of a microphone substrate same as the above. It is a circuit diagram of an impedance conversion circuit 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. (A)-(c) It is a signal waveform diagram of the signal processing part same as the above. It is a figure which shows the cancellation amount of the speaker sound by a signal processing part same as the above.

Explanation of symbols

A Communication device A1 Housing SP Speaker M1, M2 Microphone 10 Audio processing unit Ls Information line

Claims (3)

  A speaker that outputs audio information transmitted from the outside, first and second microphones that collect sound and output audio signals, and each audio signal output by the first and second microphones is signal-processed. The first microphone is disposed within a Rayleigh length determined by the area of the diaphragm of the speaker and the wavelength of the sound radiated from the diaphragm, The microphone 2 is arranged with the sound collection unit facing out of the main body, and the sound processing unit uses the sound signal output from the first microphone and the sound emitted from the speaker from the sound signal output from the second microphone. A communication device characterized by performing a process of reducing noise.   2. The communication device according to claim 1, wherein the first microphone is disposed in the main body with a sound collecting portion facing a diaphragm of the speaker.   The communication device according to claim 1, wherein the first microphone is opposed to a center of a diaphragm of the speaker.

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