JP2005260551A - Interphone system - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide an interphone system in which conventionally distorted frequency characteristics are smoothed and the frequency component of a missing sound signal is restored and corrected. <P>SOLUTION: The interphone system is composed so that an entrance slave unit 1 and a room master unit 2 are connected, a speech communication path is formed by selecting the entrance slave unit 1 from the room master unit 2, and then, the transmission/reception of the voice signal inputted from microphones 12, 28 can be executed. The interphone system is provided with a memory 23c, which stores an inverse filter coefficient prepared on the basis of the previously measured frequency characteristics of a transmitting transmission system or the receiving transmission system of the speech communication path, a Codec 23a for converting an analog signal of the sound signal to a digital signal, a voice correction digital signal processor 23 having a digital signal processor 23b which smoothes the frequency characteristics of the transmitting transmission system or the receiving transmission system after executing convolutional operation of the inverse filter coefficient to the sound signal subjected to the digital conversion, and speakers 11, 27 which output the signal subjected to the convolutional operation after converting it to the analog signal by the Codec. <P>COPYRIGHT: (C)2005,JPO&NCIPI

Description

  The present invention relates to an interphone system, and more particularly to an interphone system equipped with a sound correction digital signal processing unit.

Conventionally, as this type of intercom system, a collective entrance machine installed at a common entrance, a living room base unit installed in each dwelling unit and connected to the collective entrance machine by a call line, and another call line installed at the entrance Is known that includes an entrance slave connected to the living room master (see, for example, Patent Document 1),
Here, the living room master unit suppresses the received signal component included in the transmitted signal obtained by inputting the sound from the microphone due to the wraparound of the received signal output from the speaker, and makes a simultaneous call. Simultaneous communication means is provided, and this simultaneous communication means is used for both reception signals and transmission signals to / from the collective entrance unit and reception signals and transmission signals to the reception unit. It is comprised so that it may act.

  According to the intercom system configured as described above, the living room master unit can prevent the occurrence of howling and perform a hands-free call by two-way simultaneous call with any of the collective entrance machine and the entrance child machine. it can.

  However, in the conventional intercom system, the frequency component of the audio signal is lost due to the influence of the shape of the casing used in the system, the characteristics of the speakers and microphones used, the installation environment of the system, etc. Since the sound is distorted, there is a problem that it is impossible to provide a voice closer to the original signal input to the transmission transmission system or the reception transmission system.

JP 2000-244699 A (paragraph number “0100”, FIG. 1)

  The present invention has been made to solve such a problem, and provides an interphone system capable of flattening a frequency characteristic that has been distorted in the past and restoring and correcting a frequency component of a missing audio signal. The purpose is to do.

  The intercom system according to the first aspect of the present invention connects an entrance cordless handset and a living room master set and / or a collective entrance stand and a living room base set, respectively, and selects the entrance cordless handset or the collective entrance answering machine from the sitting room master set. An interphone system capable of transmitting and receiving voice signals input from a microphone by forming a path, and storing inverse filter coefficients created based on frequency characteristics of a transmission transmission system or a reception transmission system measured in advance on a speech path A digital signal processor for converting the analog signal of the audio signal into a digital signal, a digital signal processor for convolving the inverse filter coefficient into the digitally converted audio signal and flattening the frequency characteristics of the transmission transmission system or the reception transmission system An audio correction digital signal processing unit having a speaker that converts the convolutional signal into an analog signal by a codec and outputs the analog signal It is obtain things.

  According to a second aspect of the present invention, in the interphone system according to the first aspect of the present invention, an inverse filter coefficient created based on a frequency characteristic of a transmission system that is adapted to each of a plurality of models and measured in advance is stored in a memory. In addition, it is equipped with a connected model detection circuit that identifies the type based on the identification control signal from the entrance slave unit or the collective entrance unit connected to the living room master unit, and is adapted to the specific model detected by the connected model detection circuit. The inverse filter coefficient is selected.

  According to a third aspect of the present invention, in the interphone system according to the first aspect or the second aspect of the present invention, a ringing tone output from an entrance cordless handset or a collective entrance door connected to a living room base unit, or another The inverse filter coefficient stored in the memory is updated by the inverse filter coefficient created based on the frequency characteristic measured by the frequency characteristic measurement signal or the frequency characteristic measurement signal output from the living room master unit.

  According to a fourth aspect of the present invention, in the intercom system according to any one of the first to third aspects of the present invention, a desired frequency characteristic is stored in advance as a filter coefficient in a memory and is read out from the memory. The filter coefficient is convolved with the flattened frequency characteristic.

  According to a fifth aspect of the present invention, in the interphone system according to any one of the first to fourth aspects of the present invention, the living room base unit is a voice for a transmission transmission system to a reception transmission system with an entrance terminal. A correction digital signal processing unit is provided, and the collective entrance machine is provided with a voice correction digital signal processing unit for a transmission transmission system or a reception transmission system with each living room base unit.

  According to the intercom system of the first aspect to the fifth aspect of the present invention, the speech correction digital signal processing unit having the digital signal processor for flattening the frequency characteristics of the transmission transmission system to the reception transmission system is provided. From the first, the frequency characteristic of the interphone is flattened, and the frequency component of the voice that has been lost due to the influence of the casing or the like can be restored and corrected. Voice that is closer to the original signal input to the incoming transmission system can be provided, and thirdly, there are various cases, speakers, and microphones for intercoms. Standard sound quality can be provided by absorbing by the signal processing unit, and fourth, ideal characteristics are given to the frequency characteristics of the interphone by flattening the frequency characteristics once. Door can be.

A preferred embodiment to which an intercom system of the present invention is applied will be described below with reference to the drawings.
[Example 1]
FIG. 1 is a system configuration diagram of an intercom system when the intercom system of the present invention is applied to a detached house, and FIG. 2 is a block diagram of the intercom system in the embodiment.

  Referring to FIG. 1, the intercom system of the present invention includes an entrance cordless handset 1 installed outdoors and a living room master set 2 installed indoors. The entrance cordless handset 1 is connected via a transmission line L0. Connected to machine 2.

  As shown in FIG. 2, the entrance handset 1 has a speaker (hereinafter referred to as “slave unit speaker”) 11 and a microphone (hereinafter referred to as “slave unit microphone”) 12 for communicating with the living room base unit 2, and a guest in the room. Each part (circuit) of a call button (hereinafter referred to as a “slave unit call button”) 13 that informs the base unit 2 and a slave unit interface (hereinafter referred to as “slave unit I / F”) 14 that transmits / receives signals to / from the base unit 2. ). Here, handset I / F 14 is connected to handset speaker 11, handset microphone 12, and handset call button 13.

  The base unit 2 has a base unit interface (hereinafter referred to as “base unit I / F”) 21, a call circuit (hereinafter referred to as “base unit call circuit”) 22, and voice correction. A digital signal processing unit (hereinafter referred to as a “base unit audio correction digital signal processing unit”) 23, an operation button 24, a CPU 25, a call detection circuit 26 for detecting a call of the front door 1 and a speaker (hereinafter referred to as a “base unit speaker”). .) 27, a microphone (hereinafter referred to as “master microphone”) 28, and a connected model detection circuit 29 that detects the model of the device connected to the master I / F 21. The master audio correction digital signal processing unit 23 converts the analog signal of the audio signal into a digital signal, converts the digital signal into an audio signal (hereinafter referred to as “master codec”) 23a, and an inverse filter coefficient. A digital signal processor (hereinafter referred to as "master DSP") 23b that convolves the digitally converted voice signal to flatten the frequency characteristics of the transmission transmission system or reception transmission system, and transmission transmission measured in advance on the speech path A memory (hereinafter referred to as “parent device memory”) 23c that stores the inverse filter coefficient created based on the frequency characteristics of the transmission system or the reception transmission system. A first memory (hereinafter referred to as “first parent device memory”) 231 including a ROM or the like for storing a processing program, etc. Second memory comprising a RAM or the like to be executed by expanding the 231 program (hereinafter referred to as "second base unit memory".) And a 232. Here, the CPU 25 is connected to the base unit call circuit 22, the base unit voice correction digital signal processing unit 23, the operation button 24, the call detection circuit 26, and the connected model detection circuit 29. 27 and a parent microphone 28 are connected. The base unit I / F 21 is connected to the base unit I / F 21, the base unit call circuit 22, the call detection circuit 26, and the connected model detection circuit 29 via the transmission line L 0. Further, the base unit codec 23a of the base unit audio correction digital signal processing unit 23 is connected to the base unit call circuit 22 and the base unit DSP 23b, and the first and second base unit memories 231 and 232 are connected to the base unit DSP 23b. Has been.

This convolution calculation method will be described. Assume that a time series response hn is obtained when an impulse hn is input to the transmission system as shown in FIG. The inverse frequency characteristic H ′ (ω) is derived from the frequency characteristic H (ω) obtained by Fourier transforming the response hn (where the inverse frequency characteristic H ′ (ω) = H ⁻¹ (ω), and H (ω) · H ⁻¹ (ω) = 1). The inverse filter coefficient gn is obtained by inverse Fourier transform from the inverse frequency characteristic H ′ (ω). The input audio signal Xn to the transmission system is convolutionally calculated by digital processing with an inverse filter coefficient gn (published on July 6, 1999, first edition, Mikio Higashiyama et al., “Signal analysis and digital processing”, pp77- 80). In this way, the frequency characteristic of the transmission system is flattened.

  Next, the operation of the intercom system of the present invention configured as described above will be described.

  First, in the base unit memory 23c, a transmission / reception system or a reception unit that is measured in advance for a communication path between an entrance handset 1 and a room base unit 2 described later (hereinafter referred to as a "slave unit-base station communication path"). Inverse filter coefficients gn and g1n created based on the frequency characteristics of the transmission system are stored.

  In this state, when a visitor presses down the handset call button 13, a call signal (hereinafter referred to as a "handset call signal") is transmitted via the handset I / F 14, the transmission path L0, and the base unit I / F21. It is transmitted to the call detection circuit 26. When such a handset call signal is detected by the call detection circuit 26, the base unit call circuit 22 is controlled by the control of the CPU 25 and a ringing tone is emitted from the base unit speaker 27.

  Here, when a resident in the room who confirms the call from the visitor by the ringing sound emitted from the main unit speaker 27 picks up the handset (not shown), the hook switch (not shown) is switched, Power is supplied from a supply circuit (not shown) to the handset speaker 11 and handset microphone 12 of the entrance handset 1 via the transmission line L0. As a result, the slave unit I / F 14 between the call function of the entrance slave unit 1 (slave unit speaker 11 and slave unit microphone 12) and the call function of the room base unit 2 (master unit speaker 27 and master unit microphone 28), A communication path between the slave unit and the master unit is formed through a route to the transmission path L0, the master unit I / F 21 and the master unit call circuit 22.

  When the communication path between the slave unit and the master unit is formed in this way, the model of the entrance slave unit 1 connected to the living room master unit 2 is determined by the connected model detection circuit 29, and an identification control signal is sent from the CPU 25. It is transmitted to the base unit audio correction digital signal processing unit 23. As a result, the specific inverse filter coefficient gn adapted to the model of the entrance unit 1 is selected from the first base unit memory 231.

  In such a state, when a visitor inputs voice from the handset microphone 12, a voice signal (hereinafter referred to as "slave handset audio signal") is transmitted through the above-mentioned handset / base station communication path. The sound correction digital signal processing unit 23 is input. The slave unit audio signal is converted from an analog signal into a digital signal by the master unit codec 23a, and the digitally converted slave unit audio signal is temporarily stored in the second master unit memory 232 via the master unit DSP 23b. Here, in the master unit DSP 23b, the specific inverse filter coefficient gn adapted to the model of the entrance slave unit 1 stored in the first master unit memory 231 is digitally converted as shown in the following formula. The signal is convolved.

  However, yp: convolution calculation result (output voice), Xn: input voice, gn: inverse filter coefficient (n: 0, 1, 2, 3,...), P: current time, N: filter of g Shows the length.

  As a result, the frequency characteristic of the handset audio signal transmitted from the entrance handset 1 to the living room base unit via the handset / base unit communication path (receiving transmission system) is flattened.

  In this way, the slave unit audio signal transmitted from the entrance slave unit 1 to the living room master unit 2 is convolved with the inverse filter coefficient gn stored in the first master unit memory 231 and the master unit DSP 23b. The signal corrected and voice corrected is converted from a digital signal to an analog signal by the base unit codec 23 a and output from the base unit speaker 27 via the base unit call circuit 22.

  On the other hand, a resident's voice signal (hereinafter referred to as “master voice signal”) input from the master microphone 28 is input to the master voice correction digital signal processing unit 23 via the master call circuit 22. . The input master audio signal is converted from an analog signal to a digital signal by the master codec 23a in the same manner as described above, and the digitally converted master audio signal is sent to the second master memory 232 via the master DSP 23b. Is temporarily saved. Here, in the parent device DSP 23b, the inverse filter coefficient g1n stored in the first parent device memory 231 is convolved with the digitally converted parent device audio signal in the same manner as described above. As a result, the frequency characteristic of the master unit audio signal transmitted from the living room master unit 2 to the entrance slave unit 1 via the slave unit / base unit communication path (transmission transmission system) is flattened.

In this manner, the master unit audio signal transmitted from the room master unit 2 to the entrance slave unit 1 is convolved by the inverse filter coefficient g1n stored in the first master unit memory 231 and the master unit DSP 23b. Thus, the corrected and voice corrected signal is converted from a digital signal to an analog signal by the parent device codec 23a, and is output from the child device speaker 11 via the above-described communication path between the child device and the parent device.
[Example 2]
FIG. 3 is a system configuration diagram of the intercom system when the intercom system of the present invention is applied to a housing complex, and FIG. 4 is a block diagram of the intercom system in the embodiment.

  In FIG. 3, the intercom system according to this embodiment includes first, second, and third entrance cordless handsets 1a, 1b, and 1c installed at the entrances of the dwelling units of the apartment house, and the living rooms of the dwelling units of the apartment house. Are provided with first, second and third living room master units 2a, 2b and 2c, respectively, and a collective entrance machine 3 which is installed at the collective entrance of the apartment house. Here, the collective entrance machine 3 is connected to the first living room parent machine 2a via the first parent machine transmission line L2a, to the second living room parent machine 2b via the second parent machine transmission line L2b. 3 is connected to the third living room main unit 2c via the third main unit transmission line L2c, and the first living room main unit 2a is connected to the first entrance sub unit 1a via the first sub unit transmission line L1a. In addition, the second room master unit 2b is connected to the second entrance slave unit 1b via the second slave unit transmission line L1b, and the third room base unit 2c is connected to the second entrance unit 1b via the third slave unit transmission line L1c. 3 are connected to the three entrance cordless handsets 1c.

  Next, the collective entrance machine 3 is a collective entrance machine interface (hereinafter referred to as “collective entrance machine I / F”) 31 that transmits and receives signals to and from the first, second, and third living room master units 2a, 2b, and 2c. , A call circuit (hereinafter referred to as “collective entrance machine call circuit”) 32, a speaker (hereinafter referred to as “collective entrance machine speaker”) 33, a microphone (hereinafter referred to as “collective entrance machine microphone”) 34, a call button (hereinafter referred to as “collective entrance machine microphone”). Each unit (circuit) includes a collective entrance machine call button 35), a numeric keypad 36, and a voice correction digital signal processing unit (hereinafter referred to as a "collective entrance machine voice correction digital signal processing unit") 37. The collective entrance machine audio correction digital signal processing unit 37 converts the analog signal of the audio signal into a digital signal (hereinafter referred to as “collective entrance machine codec”) 37a, and converts the inverse filter coefficient into a digitally converted audio signal. A digital signal processor (hereinafter referred to as “collective entrance DSP”) 37b that flattens the frequency characteristics of the transmission system by performing a convolution operation, created based on the frequency characteristics of the transmission transmission system or the reception transmission system measured in advance on the speech path. The memory 37c for storing the inverse filter coefficients (hereinafter referred to as "collective entrance machine memory") 37c is provided, and the collective entrance machine memory 37c further includes a ROM for storing the inverse filter coefficients and the signal processing program. 1 memory (hereinafter referred to as “first collective entrance memory”) 371 and first collective entrance machine memory 371 And a second memory (hereinafter referred to as "second set entrance machine memory.") 372 including a RAM or the like and executing expand. Here, the collective entrance machine I / F 31 is connected to the collective entrance machine call circuit 32, the collective entrance machine call button 35 and the numeric keypad 36, and the collective entrance machine call circuit 32 includes the collective entrance machine speaker 33 and the collective entrance machine microphone 34. Is connected. The collective entrance machine codec 37a of the collective entrance machine audio correction digital signal processing unit 37 is connected to the collective entrance machine call circuit 32 and the collective entrance machine DSP 37b, and the collective entrance machine DSP 37b includes first and second collective entrance machines. Memories 371 and 372 are connected. Furthermore, the collective entrance unit I / F 31 is connected to the first room base unit 2a via the first base unit transmission path L2a, and to the second room base unit 2b via the second base unit transmission path L2b. The third base unit transmission line L2c is connected to the third room base unit 2c.

  In addition, the 1st, 2nd, 3rd entrance cordless handset 1a, 1b, 1c and the 1st, 2nd, 3rd room main unit 2a, 2b, 2c are the entrance cordless handset 1 and the room shown in FIG. Since it is the same as that of the base unit 2, a common reference numeral is attached and detailed description is omitted.

  Next, the operation of the intercom system according to the second embodiment configured as described above will be described.

  First, in the collective entrance machine memory 37c, communication paths (hereinafter referred to as "first, second, and third" between the collective entrance machine 3, first, second, and third room master units 2a, 2b, and 2c, which will be described later, are provided. The inverse filter coefficient g3n created on the basis of the frequency characteristics of the transmission transmission system or the reception transmission system measured in advance is stored for the first entrance main unit 2a. In the master unit memory 23c, a second room master unit for the communication path between the first room master unit 2a and the collective entrance unit 3 (hereinafter referred to as "first collective entrance unit / base unit call path") is stored. In the base unit memory 23c of 2b, there is a third room for the communication path between the second room base unit 2b and the collective entrance 3 (hereinafter referred to as "second collective entrance / base unit call path"). In the base unit memory 23c of the base unit 2c, a communication path between the third room base unit 2c and the collective entrance 3 (hereinafter referred to as “third collective entrance”). For machine-master unit between speech path "hereinafter.), The inverse filter coefficient g2n) created based on the frequency characteristic of the pre-measured transmission transmission system to the receiving transmission system respectively are stored.

  In such a state, for example, a visitor calls the resident in the dwelling unit (hereinafter referred to as “first resident”) related to the first living room master unit 2a, so that the numeric keypad 36 of the collective entrance 3 is called. When the room number of the first living room master unit 2a is selected and the collective entrance machine call button 35 is pressed, a call signal (hereinafter referred to as "collective entrance machine call signal") is sent to the collective entrance machine I / F 31, the first A call detection circuit for the first living room parent device 2a via the parent device transmission line L2a and the parent device I / F (hereinafter referred to as “first parent device I / F”) 21 of the first living room parent device 2a ( Hereinafter, it is referred to as “first call detection circuit”) 26. When the collective entrance machine call signal is detected by the first call detection circuit 26, the first room master unit is controlled by a CPU (hereinafter referred to as "first CPU") 25 of the first room base unit 2a. The master phone call circuit 2a (hereinafter referred to as “first master phone call circuit”) 22a is controlled and the master speaker of the first living room master device 2a (hereinafter referred to as “first master phone speaker”). A ringing tone is emitted from 27.

  Here, when the first resident operates a call button (not shown) of the first room base unit 2a based on the ringing sound emitted from the first base unit speaker 27, the collective entrance 3 Call function (collective entrance speaker 33 and collective entrance microphone 34) and call function (first master speaker 27 and master microphone of first living room master 2a) 28), the central entrance I / F 31, the first master unit transmission path L2a, the first master unit I / F 21 and the first master unit call circuit 22 A first communication path between the main entrance machine and the parent machine is formed through the route to reach.

  When the first collective entrance / base station communication path is formed in this way, the model of the collective entrance 3 connected to the first living room master 2a is determined by the connection model detection circuit 29. Then, the identification control signal is transmitted from the first CPU 25 to the base unit voice correction digital signal processing unit (hereinafter referred to as “first base unit voice correction digital signal processing unit”) 23 of the first room base unit 2a. As a result, a specific inverse filter coefficient g2n adapted to the model of the collective entrance 3 is selected from the first memory 231 of the first living room master 2a.

  In this state, when a visitor inputs a voice from the collective entrance microphone 34, a voice signal (hereinafter referred to as “collective entrance voice signal”) is transmitted through the first collective entrance / base unit communication path. Is input to the first master audio correction digital signal processing unit 23. The central entrance audio signal is converted from an analog signal into a digital signal by a master codec (hereinafter referred to as “first master codec”) 23a of the first master audio correction digital signal processing unit 23, and converted into a digital signal. The collected entrance door audio signal is sent to the first master unit voice correction digital signal via the master unit DSP (hereinafter referred to as “first master unit DSP”) 23 b of the first master unit voice correction digital signal processing unit 23. Temporarily stored in the second master memory 232 of the processing unit 23. Here, in the master unit DSP 23b of the first master unit audio correction digital signal processing unit 23, the specific inverse filter coefficient g2n adapted to the type of the collective entrance unit 3 stored in the first master unit memory 231 is described above. The convolution operation is performed on the voice signal of the collective entrance machine that has been digitally converted. As a result, the frequency characteristic of the collective entrance unit audio signal transmitted from the collective entrance unit 3 to the first living room base unit 2a through the first collective entrance unit / base unit call (receiving transmission system) is flattened. The

  In this way, the collective entrance unit audio signal transmitted from the collective entrance unit 3 to the first living room base unit 2a is the reverse stored in the first base unit memory 231 of the first living room base unit 2a. The signal corrected by the convolution calculation by the filter coefficient g2n and the parent device DSP 23b of the first room parent device 2a and the sound corrected is converted from a digital signal to an analog signal by the first parent device codec 23a. Is output from the first parent speaker 27 via the parent device call circuit 22.

  On the other hand, the voice signal of the first resident (hereinafter referred to as “first master unit voice signal”) input from the first master unit microphone 28 is sent to the collective entrance unit 3 by the connected model detection circuit 29. Since the presence of the machine audio correction digital signal processing unit 37 is known, the call is made through the first collective entrance / base phone call without passing through the first machine audio correction digital signal processing unit 23. Then, it is input to the collective entrance machine voice correction digital signal processing unit 37. The input first master unit audio signal is converted from an analog signal to a digital signal by the collective entrance unit codec 37a in the same manner as the process in the first room master unit 2a described above, and the first master unit subjected to digital conversion is converted into a digital signal. The audio signal is temporarily stored in the collective entrance machine memory 372 via the collective entrance machine DSP 37b. Here, in the collective entrance machine DSP 37b, as described above, the inverse filter coefficient g3n stored in the first collective entrance machine memory 371 is convolved with the first base unit audio signal digitally converted as described above. Is done. As a result, the frequency characteristic of the first master unit audio signal transmitted from the first room master unit 2a to the collective entrance unit 3 via the first collective entrance unit / base unit call (transmission transmission system) is obtained. Flattened.

  In this way, the first master unit audio signal transmitted from the first room master unit 2a to the collective entrance unit 3 is combined with the inverse filter coefficient g3n stored in the first collective entrance unit memory 371. The signal corrected and convoluted by the convolution operation by the entrance machine DSP 37b is converted from a digital signal to an analog signal by the collective entrance machine codec 37a, and is sent from the collective entrance machine peaker 33 via the collective entrance machine call circuit 32. Is output.

  The intercom system of the present invention is not limited to the above-described embodiment, and, for example, the same effects as the intercom system of the present invention can be obtained as follows.

  First, the inverse filter coefficients created on the basis of the frequency characteristics of the transmission system, which are adapted to each of a plurality of models and measured in advance, are stored in the memories 23c and 37c, and the entrance cordless devices 1, 1a, 1b, 1c, A specific inverse filter coefficient adapted to a specific model detected by the connected model detection circuit 29 may be selected by an identification control signal from the collective entrance 3.

  Secondly, the ringing tone output from the entrance unit 1, 1a, 1b, 1c or the collective entrance unit 3, or another frequency characteristic measurement signal, or the operation buttons 24 of the room base units 2, 2a, 2b, 2c. The inverse filter coefficients stored in the memories 23c and 37c may be updated with the inverse filter coefficients created based on the frequency characteristics measured by the frequency characteristic measurement signal output by the pressing operation such as.

  Third, a desired frequency characteristic (for example, an ideal frequency characteristic required for the product) is stored in advance in the memories 23c and 37c as a filter coefficient, and the filter coefficient read from the memories 23c and 37c is the desired frequency characteristic. It may be configured to perform a convolution operation.

  Fourthly, the number of the entrance slave units 1, 1a, 1b, 1c, the living room master units 2, 2a, 2b, 2c, and the collective entrance unit 3 is not limited to those in the embodiment, and the number may be increased or decreased depending on the scale of the system. can do.

The system block diagram of the intercom system in 1st Example of this invention. The block diagram of the intercom system in the 1st example of the present invention. The system block diagram of the intercom system in 2nd Example of this invention. The block diagram of the intercom system in 2nd Example of this invention. Explanatory drawing about the convolution calculation method used for the intercom system of this invention.

Explanation of symbols

1 ... Entrance cordless handset
11 ... Speaker (Slave unit speaker)
12 ... Microphone (Slave unit microphone)
2 ... Living room master unit 23 ... Audio correction digital signal processing unit 23a ... Codec 23b ... Digital signal processor (DSP)
23c ... Memory 27 ... Speaker (master speaker)
28 ... Microphone (main unit microphone)
29 ... Connected model detection circuit 3 ... Collective entrance machine 33 ... Speaker (collective entrance machine speaker)
34 ... Microphone (collector entrance microphone)
Xn: Audio signal gn, g1n, g2n, g3n: Inverse filter coefficient

Claims (5)

Connect the entrance slave unit (1) and the living room master unit (2) and / or the collective entrance unit (3) and the living room master unit, and select the entrance slave unit or the collective entrance unit from the living room master unit. An intercom system capable of transmitting and receiving voice signals input from microphones (12, 28, 34) by forming a path
A memory (23c, 37c) for storing an inverse filter coefficient created based on a frequency characteristic of a transmission transmission system or a reception transmission system measured in advance on the speech path, and a codec for converting an analog signal of the audio signal into a digital signal ( 23a, 37a), audio correction having a digital signal processor (23b, 37b) that convolves the inverse filter coefficients with the digitally converted audio signal to flatten the frequency characteristics of the transmission transmission system or reception transmission system An interphone system comprising: a digital signal processing unit (23, 37); and a speaker (11, 27, 33) that converts a signal subjected to convolution operation into an analog signal by the codec and outputs the analog signal. An inverse filter coefficient created based on a frequency characteristic of a transmission system that is measured in advance and adapted to each of a plurality of models is stored in the memory, and the entrance slave unit or the collective entrance connected to the living room master unit A connection model detection circuit (29) for identifying the type by an identification control signal from the machine,
2. The intercom system according to claim 1, wherein a specific inverse filter coefficient adapted to a specific model detected by the connected model detection circuit is selected.   Measured by the ringing tone output from the entrance slave unit or the collective entrance unit connected to the living room base unit, another frequency characteristic measurement signal, or the frequency characteristic measurement signal output from the room base unit The interphone system according to claim 1 or 2, wherein the inverse filter coefficient stored in the memory is updated with an inverse filter coefficient created based on a frequency characteristic.   4. A desired frequency characteristic is stored in advance as a filter coefficient in the memory, and the filter coefficient read from the memory is convolved with the flattened frequency characteristic. The intercom system according to claim 1. The living room master unit is provided with a voice correction digital signal processing unit for a transmission transmission system to a reception transmission system with the entrance slave unit,
The intercom system according to any one of claims 1 to 4, wherein the collective entrance machine is provided with a voice correction digital signal processing unit for a transmission transmission system or a reception transmission system with respect to each parent device. .

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