JP2007124163A - Call apparatus - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a call apparatus capable of conducting a pleasant call, even when performing switching between a handset call and an amplified call. <P>SOLUTION: When a call mode is switched to a handset call mode by a call mode switching section 20, a call transmission side filter 21 and a call reception side filter 22 are activated to filter a call transmission signal and a call reception signal. As a result, since the difference between the frequency characteristics of a voice signal in an amplified call mode and that in the handset call mode is corrected by the call transmission side filter 21 and the call reception side filter 22 and the voice quality of the call voice, even when performing switching between the handset call mode and the amplified call mode, a sense of incongruity and a sense of discomfort will not given to the talker, and the talker can conduct a pleasant call. <P>COPYRIGHT: (C)2007,JPO&INPIT

Description

  The present invention relates to a call device such as an interphone that performs both a loudspeaker call using a microphone and a speaker and a call using a handset.

Conventionally, a user can selectively switch between a call using a handset (handset call) and a call using a microphone and a speaker provided in the communication device main body instead of the handset (amplified call). A telephone device is provided (see, for example, Patent Document 1).
Japanese Utility Model Publication No. 6-48260

  However, if the input characteristics (frequency characteristics) of the microphones and speakers built into the handset differ from the input characteristics (frequency characteristics) of the microphones and speakers built into the main body of the communication device, the handset call and the voice call are switched. At times, the sound quality may change, causing the speaker to feel uncomfortable or uncomfortable.

  The present invention has been made in view of the above circumstances, and an object of the present invention is to provide a call device capable of making a comfortable call even when a handset call and a loud call are switched.

  In order to achieve the above object, the first aspect of the present invention provides a microphone for outputting collected sound as a transmission signal provided in the communication device main body, and a reception signal from the other communication terminal provided in the communication device main body. A speaker that rings in response to the voice signal, a receiver signal path through which a reception signal transmitted from the other party's telephone terminal is transmitted, and a transmitter signal path through which a transmission signal to be transmitted to the other party's telephone terminal is transmitted. , A handset provided separately from the main body of the communication device, a speech communication mode for connecting the receiver side signal path and the transmitter side signal path to the speaker and the microphone, and a handset for connecting the receiver side signal path and the transmitter side signal path to the handset Call mode switching means for selectively switching between call modes, frequency characteristics of voice signals in the voice call mode, and voice signals in the handset call mode Characterized by comprising a frequency characteristic correction means for correcting the difference in frequency characteristics.

  According to a second aspect of the present invention, in the first aspect of the present invention, in the handset call mode, in the handset call mode, a side sound generating means for generating a side sound by connecting between the transmission side signal path and the reception side signal path, And an echo canceller that cancels the line echo caused by the signal sneak in the line between the acoustic coupling and the partner telephone terminal. The echo canceller adaptively identifies the impulse response of the line echo path and An adaptive filter for estimating a pseudo echo component from an input signal to the path; and a subtractor for subtracting the pseudo echo component estimated by the adaptive filter from an output signal from the line echo path. The signal level of the side sound signal output to the side signal path is increased or decreased according to the pseudo echo component estimated by the adaptive filter.

  According to a third aspect of the present invention, in the first aspect of the present invention, the receiving mode and the transmitting mode are set by inserting a loss into the receiving side signal path and the transmitting side signal path and adjusting an amount of loss inserted into each signal path. A voice switch for switching and a recording means for recording at least the received signal are provided, and the recording means records the received signal before the loss is inserted by the voice switch.

  According to a fourth aspect of the present invention, in the first aspect of the present invention, the specific sound generating means for generating a specific sound as an excuse for canceling the call by letting the other party talk, and the generation of the specific sound without using the call mode switching means And a signal path for sending a specific sound from the means to the speaker.

  According to the first aspect of the present invention, since the frequency characteristic correcting means corrects the difference between the frequency characteristic of the voice signal in the voice call mode and the frequency characteristic of the voice signal in the handset call mode, the handset call mode and the voice call mode are switched. Even when the sound quality of the call voice does not change, it is possible to make a comfortable call without causing the speaker to feel uncomfortable or uncomfortable.

  According to the invention of claim 2, the signal level of the side sound signal generated by the side sound generating means is increased or decreased according to the increase or decrease of the signal level of the received signal by the echo canceller. You can make comfortable calls.

  According to the invention of claim 3, since the recording means records the received signal before the loss is inserted by the voice switch, there is a difference in the recording volume between the voice recorded by the handset call and the voice recorded by the voice call. Does not occur.

  According to the invention of claim 4, for example, when talking to the other party's call terminal, an excuse to stop the call by making the other party hear a specific sound such as a ringing tone of a telephone or a child's cry, You can quickly cancel a call with a party you do not want to call. In addition, when a specific sound is emitted from the speaker provided on the main body of the communication device, a specific sound with echo and noise is generated from the other party's call terminal. In comparison, a specific sound can be heard authentically.

  Hereinafter, an embodiment in which the technical idea of the present invention is applied to a dwelling unit installed in each dwelling unit in an intercom system for an apartment house (such as an intercom master unit or a housing information panel) will be described in detail with reference to the drawings.

  In the present embodiment, as shown in FIG. 1, a microphone 1 that is provided in a communication device main body (not shown) and outputs the collected sound as a transmission signal, and a communication terminal provided in the communication device main body from a counterpart communication terminal. A speaker 2 that rings according to a received signal, a received signal path through which a received signal transmitted from the other party's telephone terminal is transmitted, and a transmitting side signal through which a transmitted signal to be transmitted to the other party's telephone terminal is transmitted. A two-wire / four-wire converter 3 for transmitting a route to / from speech lines L1, L2, and L3, a handset 4 provided separately from the speech device main body, a receiver signal route, and transmission A call mode switching unit 20 for selectively switching between a loudspeaking call mode in which the side signal path is connected to the speaker 2 and the microphone 1 and a handset call mode in which the receiving side signal path and the transmitting side signal path are connected to the handset 4; And a frequency characteristic correcting means serving transmitting side filter 21 and the receiving side filter 22 for correcting a difference in frequency characteristic of the audio signal in the frequency characteristic and the handset communication mode of the audio signal in the speaker-phone call mode.

  Three lines of call lines L 1, L 2, L 3 are connected to the 2-wire-4 line conversion unit 3 via a call line switching unit 50. A door phone handset (not shown) installed at the exterior entrance of each dwelling unit is connected to the first call line L1, and another dwelling unit installed in the same dwelling unit is connected to the second call line L2. And a third telephone line L3 is connected to a lobby interphone (not shown) installed at the common entrance and an alarm monitoring board (not shown) installed in the management room. ) Or the like, and any one of the call lines L1 to L3 and the 2-wire to 4-wire conversion unit 3 are alternatively switched and connected by the call line switching unit 50. That is, if the first call line L1 is connected, a call can be made with the intercom unit, and if the second call line L2 is connected, a call can be made with another dwelling unit in the house. If the third call line L3 is connected, a call can be made between the lobby intercom and the alarm monitoring panel.

  The call mode switching unit 20 is composed of an analog switch, and the contact 20a of the receiver signal path is connected to a contact 20b connected to the speaker 2 or a speaker (not shown) built in the receiver of the handset 4. The switch 20c is alternatively switched and connected, and a switching operation is performed in accordance with an operation of an operation button (not shown) provided on the communication device body.

  Further, the present embodiment is composed of a DSP or CPU equipped with software and converts a signal processing unit M that performs signal processing for a call and an analog transmission signal collected by the microphone 1 into a digital transmission signal. An A / D converter 5, a D / A converter 6 that converts a digital reception signal output from the signal processing unit M into an analog reception signal and outputs the analog reception signal to the call mode switching unit 20, and a mouthpiece of the handset 4 An A / D converter 9 that converts an analog transmission signal collected by a built-in microphone (not shown) into a digital transmission signal, and an analog reception signal transmitted through signal lines L1, L2, and L3 An A / D converter 8 that converts a signal into a digital received signal, and a digital transmitted signal output from the signal processing unit M is converted into an analog transmitted signal. And a D / A converter 7 that.

  The signal processing unit M includes a transmission side filter 21 and a reception side filter 22, first and second echo cancellers 30 </ b> A and 30 </ b> B, a voice switch 10, and a side sound generation unit 40. First, the voice switch 10 and the first and second echo cancellers 30A and 30B will be described in more detail with reference to FIGS. However, in FIG. 2, components unnecessary for the description of the transmitting side filter 21 and the receiving side filter 22 are not shown.

The first echo canceller 30A includes a well-known structure composed of the adaptive filter 31A and a subtractor 32A, the impulse response of the feedback path (acoustic echo path) H _AC formed by the acoustic coupling between the speaker 2 microphone 1 Acoustic echo is suppressed by subtracting the pseudo echo component (acoustic echo) that is adaptively identified by the adaptive filter 31A and estimated from the reference signal (input signal to the speaker 2) from the output signal of the microphone 1 by the subtractor 32A. Is. The second echo canceller 30B also has a conventionally known configuration including an adaptive filter 31B and a subtractor 32B. The reflection due to impedance mismatch between the two-wire / four-wire converter 3 and the transmission path and the counterpart An impulse response of a feedback path (line echo path) H _LIN formed by acoustic coupling between a speaker and a microphone in a communication terminal (door phone slave unit, etc.) is adaptively identified by the adaptive filter 31B, and a reference signal (2-wire- The line echo is suppressed by subtracting the pseudo echo component (line echo) estimated from the input signal to the 4-wire conversion unit 3, that is, the transmission signal) from the reception signal by the subtractor 32B.

The voice switch 10 includes a transmission side loss insertion unit 11 that inserts a loss into the signal path on the transmission side, a reception side loss insertion unit 12 that inserts a loss into the signal path on the reception side, and a transmission side and a reception side. And an insertion loss amount control unit 13 that controls the amount of loss inserted from each of the loss insertion units 11 and 12. The insertion loss amount control unit 13, the path to return to the input point Tin of the transmitting end losses insertion portion 11 from the output point Rout of the receiving-side loss insertion portion 12 via the acoustic echo path H _AC (hereinafter, "sound side feedback path )) And a return path from the output point Tout of the transmission side loss insertion unit 11 to the input point Rin of the reception side loss insertion unit 12 via the line echo path H _LIN (hereinafter referred to as “reception side feedback gain α”). (Referred to as “line-side feedback path”), and the loss amount to be inserted into the closed loop is estimated based on the estimated values α ′ and β ′ of the feedback gains α and β on the acoustic side and the line side. A total loss amount calculation unit 14 that calculates the sum (the sum of the insertion loss amount of the transmission side loss insertion unit 11 and the insertion loss amount of the reception side loss insertion unit 12), and the call state by monitoring the transmission signal and the reception signal According to the estimation result and the value calculated by the total loss calculation unit 14 And an insertion loss amount distribution processing unit 15 that determines the distribution of each insertion loss amount of the side loss insertion unit 11 and the reception side loss insertion unit 12. As described above, the first and second echo cancellers 30A and 30B and the voice switch 10 in the present embodiment are realized by controlling the DSP hardware with software (programs) for the echo canceller and the voice switch. Has been. Therefore, the input / output signals (received signal and transmitted signal) of the voice switch 10 and the first and second echo cancellers 30A and 30B in the following description are sampled at a predetermined sampling period, and the A / D converters 5 and 8 are used. It is quantized by.

The total loss amount calculation unit 14 estimates the time-average power of the input signal of the transmission side loss insertion unit 11 in a short time using a rectifier / smoothing device, a low-pass filter, and the like. estimating the time average power in a short time of the output signal of the receiving-side loss insertion portion 12 with a like, time of the output signal of the receiving-side loss insertion portion 12 in the maximum delay time assumed in acoustic side feedback path H _AC A minimum value of the estimated value of the average power is obtained, and a value obtained by dividing the estimated value of the time average power of the input signal of the transmission side loss insertion unit 11 by this minimum value is set as the estimated value α ′ of the acoustic feedback gain α. The time average power of the input signal of the receiving side loss insertion unit 12 in a short time is estimated using a rectifying / smoothing device, a low-pass filter, etc. Insertion part 11 Estimating the time average power in a short time of the signal, determining the minimum value of the estimated value of the time average power of the output signal of the transmitter-side loss insertion unit 11 at the maximum delay time assumed in the line side feedback path H _LIN, A value obtained by dividing the estimated value of the time average power of the input signal of the receiving side loss insertion unit 12 by this minimum value is defined as an estimated value β ′ of the line side feedback gain β. Then, the total loss calculation unit 14 calculates a total loss Lt necessary to obtain a desired gain margin MG from the estimated values α ′ and β ′ of the acoustic feedback gain α and the line feedback gain β. The value Lt is output to the insertion loss amount distribution processing unit 15.

  The insertion loss amount distribution processing unit 15 monitors the input / output signals of the transmission side loss insertion unit 11 and the input / output signals of the reception side loss insertion unit 12, and compares the power levels of these signals and the presence / absence of a voice signal. The communication state (the reception state, the transmission state, etc.) is determined from the information of the information, and the total loss Lt is distributed to the transmission side loss insertion unit 11 and the reception side loss insertion unit 12 at a rate corresponding to the determined communication state. The insertion loss amount of each loss insertion part 11 and 12 is adjusted so that it may.

  By the way, as described above, the total loss amount calculation unit 14 according to the present embodiment calculates and adaptively updates the sum of loss amounts to be inserted into the closed loop based on the estimated values α ′ and β ′ of the feedback gains α and β. There are two operation modes, an update mode and a fixed mode for fixing the total loss amount to a predetermined initial value, and the first and second echo cancellers 30A and 30B are sufficiently provided from the start of a call with the other party's call terminal. It operates in the fixed mode during the period until convergence, and operates in the update mode during the period after the first and second echo cancellers 30A and 30B have sufficiently converged. That is, in the total loss amount calculation unit 14, the estimated values α ′ and β ′ of the acoustic side feedback gain α and the line side feedback gain β are continuously maintained for a predetermined time (several hundred milliseconds) for a predetermined threshold value ε ( For example, it is considered that the first and second echo cancellers 30A and 30B have sufficiently converged when the values are less than 10 dB to 15 dB smaller than the estimated values α ′ and β ′ at the start of the call, Before, the operation mode is switched to the update mode in which the total loss amount is adaptively updated based on the estimated values α ′ and β ′. Note that the initial value of the total loss amount in the fixed mode is set to a value sufficiently larger than the total loss amount updated as needed in the update mode.

  Thus, when the first and second echo cancellers 30A and 30B immediately after the start of the call are not sufficiently converged, the total loss amount calculation unit 14 operating in the fixed mode sets the value sufficiently large. Since the initial total loss amount is inserted into the closed loop, it is possible to suppress the generation of unpleasant echoes (acoustic echoes and line echoes) and howling, and realize a stable half-duplex call. In the state where the first and second echo cancellers 30A and 30B have sufficiently converged after the time from the start of the call, the operation mode of the total loss calculation unit 14 is switched from the fixed mode to the update mode and closed loop. Since the total loss amount to be inserted into the value decreases to a value sufficiently lower than the initial value, two-way simultaneous calls can be realized.

  Here, a specific operation of the total loss amount calculation unit 14 in the update mode will be described with reference to a flowchart of FIG.

  The total loss amount calculation unit 14 executes an estimation process of the acoustic side feedback gain α and the line side feedback gain β at a predetermined sampling period from the time when the fixed mode is changed to the update mode (t = t1), and the estimated value α '(n), β' (n) is calculated (step 1), and the gain margin of the closed loop MG [is calculated from the product of these two estimated values α '(n), β' (n) and the gain margin MG. The desired total loss amount Lr (n) required for maintaining the value [dB] is calculated by the following equation (step 2).

Lr (n) = 20 log | α ′ (n) · β ′ (n) | + MG [dB]
Note that α ′ (n), β ′ (n), and Lr (n) indicate an estimated value of feedback gain and a desired total loss amount calculated by sampling n times from the update mode transition point, respectively. Further, the total loss amount calculation unit 14 calculates the n-th total loss amount desired value Lr (n) calculated from the above formula and the previous (n−1) th total loss amount Lt (n−1), that is, the previous process. When the desired total loss amount Lr (n) calculated this time is larger than the total loss amount determined and actually inserted, a slight increase Δi [dB in the previous total loss amount Lt (n−1). ] Is defined as the total loss amount Lt (n) = Lt (n−1) + Δi (steps 3 and 4), and the total loss calculated this time with respect to the previous total loss amount Lt (n−1). When the loss desired value Lr (n) is small, the current total loss Lt (n) = Lt (n) is obtained by subtracting a slight decrease Δd [dB] from the previous total loss Lt (n−1). −1) −Δd (steps 5 and 6).

  Thus, by suppressing the increase / decrease in the total loss amount by the total loss amount calculation unit 14 to a small value of Δi or Δd, the first and second echo cancellers 30A can be used just after the start of a call with the other party's call terminal. , 30B actively update the coefficient toward convergence, so that a sense of incongruity can be eliminated even when the acoustic feedback gain α and the line feedback gain β change significantly.

Next, the transmitting side filter 21 and the receiving side filter 22 will be described. 4A shows the input characteristics (frequency characteristics) H ^ of the microphone 1, and FIG. 4B shows the input characteristics (frequency characteristics) H 'of the microphone built in the handset 4, respectively. Are X and X ', and outputs are Y and Y', the relationship of Y = H ^ .X and Y '= H ^'. X 'is established. Therefore, in order to make the output Y of the microphone 1 equal to the output Y ′ of the microphone incorporated in the handset 4, the filter coefficient Z (= H ^ ′ · H ^ ⁻¹ , where Y ′ = Z · Y, FIG. It is only necessary to filter the output Y ′ of the microphone built in the handset 4 with a filter (sending side filter 21) having (see (c))). Similarly, the input to the speaker built in the handset 4 is filtered by a filter (receiving-side filter 22) having a filter coefficient obtained by multiplying the input characteristic of the speaker built in the handset 4 by the inverse characteristic of the input characteristic of the speaker 2. Then, the output of the speaker 2 and the output of the speaker built in the handset 4 can be made equal.

  Thus, in the present embodiment, when the call mode switching unit 20 is switched to the handset call mode, the transmission side filter 21 and the reception side filter 22 having the above filter coefficients are operated to transmit the transmission signal and the reception line. By filtering the signal, the difference between the frequency characteristics of the voice signal in the loudspeaker call mode and the frequency characteristic of the voice signal in the handset call mode is corrected, so that the voice quality of the call voice is improved even when the handset call mode and the loudspeaker call mode are switched. Since it does not change, the speaker does not feel uncomfortable or uncomfortable, and a comfortable call can be made. However, when the call mode switching unit 20 is switched to the loudspeaker call mode, it is necessary to stop the operations of the transmission side filter 21 and the reception side filter 22 to allow the transmission signal and the reception signal to pass through.

  Next, the side sound generator 40 will be described. The side sound generation unit 40 includes a filter 41 that passes only the frequency band component of the voice from the transmission signal that is output from the first echo canceller 30A and is input to the voice switch 10, and the transmission signal that has passed through the filter 41 (hereinafter referred to as the transmission signal). The side sound signal amplified by the variable amplifier 42 is superimposed on the reception signal output from the voice switch 10. Here, since the signal level of the reception signal output from the voice switch 10 increases or decreases according to the amount of suppression of the line echo by the second echo canceller 30B, the relative signal level of the side sound signal with respect to the reception signal also increases or decreases. As a result, the volume of the side sound audible from the speaker 2 or the handset 4 may be increased or decreased more than necessary, which may cause the speaker to feel uncomfortable or uncomfortable. Therefore, in the present embodiment, the received sound is adjusted by adjusting the amplification degree of the variable amplifier 42 in the side sound generating unit 40 according to the level of the pseudo echo component subtracted from the received signal by the subtractor 32B of the second echo canceller 30B. The volume change of the side sound with respect to the voice is suppressed so that a comfortable call can be performed.

  By the way, in the present embodiment, a recording / reproduction processing unit 60 for recording a call signal (a reception signal and a transmission signal), reproducing the recorded call signal, and causing the speaker 2 or the speaker built in the handset 4 to ring. In addition, a memory unit 61 is provided. The recording / playback processing unit 60 compresses the call signal input from the transmission side signal path and the reception side signal path into digital data by ADPCM (Adaptive Differential Pulse Code Moudulation), and writes the compressed call data to the memory unit 61. In addition to storing (recording), a call signal obtained by decompressing and restoring the call data read out from the memory unit 61 is output to the receiver signal path. Here, in the recording / playback processing unit 60, by recording the received signal and the transmitted signal before the loss is inserted by the voice switch 10, between the voice recorded by the handset call and the voice recorded by the voice call. In order to prevent differences in recording volume during playback.

  In addition, the present embodiment includes a specific sound generator 70 that generates a specific sound as an excuse for stopping the call by letting the other party call. For example, when a resident responds to a call from the lobby intercom and the other party (calling party) is an uninvited visitor, such as a salesman, he / she wants to cancel the call as soon as possible and collect it from the visitor. And home-stayers may want. In such a case, you can create an excuse to stop the call by letting the other party hear a sound such as a telephone ringing tone or a child's cry (named "specific sound"). For example, "The phone is ringing now You can quickly cancel a call with a person you do not want to talk to, such as "Please pick up." Or "Please pick up because the child is crying. The specific sound as described above is compressed and stored as digital data in a memory (not shown), and when the speaker (at-home person) operates an operation switch provided on the body of the communication device, the specific sound generator 70 is provided. The digital data read from the memory is decompressed, converted into an analog signal, and output to the speaker 2. Thus, when a specific sound is emitted from the speaker 2, a specific sound accompanied by an echo or noise in the house is sounded from the other party's call terminal (lobby intercom, etc.). There is an advantage that a specific sound can be heard in a genuine manner as compared with the case of directly superimposing.

It is a block diagram which shows embodiment of this invention. It is a block diagram which shows the audio | voice switch and 1st and 2nd echo canceller in the same as the above. It is a flowchart for operation | movement description of a voice switch in the same as the above. (A)-(c) is operation | movement explanatory drawing of the transmission side filter in the same as the above.

Explanation of symbols

DESCRIPTION OF SYMBOLS 1 Microphone 2 Speaker 4 Handset 10 Voice switch 20 Call mode switching part 21 Transmission side filter 22 Reception side filter 30A 1st echo canceller 30B 2nd echo canceller

Claims (4)

  A microphone that is provided in the main body of the communication device and outputs the collected sound as a transmission signal, a speaker that is provided in the main body of the communication device and that rings in response to a reception signal from the other party's telephone terminal, and is transmitted from the other party's telephone terminal. Receiving side signal path through which the incoming call signal is transmitted, transmitting side signal path through which the transmission signal to be sent to the other party's telephone terminal is transmitted, a handset provided separately from the telephone apparatus main body, and the receiving side A call mode switching means for selectively switching between a voice call mode for connecting the signal path and the transmission side signal path to the speaker and the microphone and a handset call mode for connecting the reception side signal path and the transmission side signal path to the handset; Frequency characteristic correction means for correcting the difference between the frequency characteristic of the voice signal in the voice call mode and the frequency characteristic of the voice signal in the handset call mode Call device characterized by comprising a. In the handset call mode, on the line between the side-tone generating means for connecting the transmitting-side signal path and the receiving-side signal path to generate a side sound, and the acoustic coupling at the other party's call terminal and the other party's call terminal It has an echo canceller that cancels line echo caused by signal wraparound,
The echo canceller adaptively identifies the impulse response of the line echo path and estimates the pseudo echo component from the input signal to the line echo path, and the pseudo echo component estimated by the adaptive filter from the line echo path. A subtractor for subtracting from the output signal of
2. The communication apparatus according to claim 1, wherein the side sound generating means increases or decreases the signal level of the side sound signal output to the receiver side signal path according to the pseudo echo component estimated by the adaptive filter. A voice switch for switching between a reception mode and a transmission mode by inserting a loss into the reception side signal path and a transmission side signal path and adjusting a loss amount to be inserted into each signal path; and a recording means for recording at least the reception signal; With
2. The call device according to claim 1, wherein said recording means records a received signal before loss is inserted by a voice switch.   A specific sound generating means for generating a specific sound as an excuse for letting the other party hear the call and a signal path for sending the specific sound from the specific sound generating means to the speaker without using the call mode switching means The call device according to claim 1.

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