JP4900184B2 - Loudspeaker - Google Patents

Description

  The present invention relates to a loudspeaker device (interphone, telephone, PHS, etc.) used in a house, office, factory or the like.

  Conventionally, it is not necessary to have a handset during a call, and a voice signal transmitted from the other party's call terminal is sent to the caller away from the call terminal by a speaker, and the voice emitted by the caller is microphone A loudspeaker device is provided that enables half-duplex calling by collecting sound and transmitting it to the other party's call terminal. In such a loudspeaker apparatus, a two-wire / four-wire conversion hybrid circuit required when acoustic coupling between a speaker and a microphone, which are constituent elements thereof, and a transmission path of an audio signal are configured in a two-wire configuration. A closed loop is formed on the speech path due to sneaking from the transmission signal path to the reception signal path caused by impedance mismatch in the voice, and acoustic coupling between the speaker and the microphone in the other party's speech terminal. If it becomes more than twice, howling occurs, and if howling occurs, the call cannot be continued, and means for suppressing this is required.

  Therefore, in the conventional loudspeaker apparatus, it is determined whether the call state is the reception state or the transmission state by monitoring the transmission signal and the reception signal, and the transmission signal path is determined according to the determined communication state. Alternatively, a voice switch has been widely used in which attenuating means is inserted in at least one of the reception signal paths to reduce the closed loop loop gain and prevent howling. The basic operation of the voice switch is to estimate the power of the transmission signal and the reception signal, compare the magnitude relationship between them, and insert a predetermined amount of loss to the side with the smaller instantaneous power. Further, when use in an environment with a high level of ambient noise is expected, a means for determining whether the transmitted signal and the received signal are voice or non-voice is required. Furthermore, the acoustic coupling gain and line feedback gain when viewed from the voice switch are large, and the transmission signal and the reception signal include not only the near-end voice signal and the far-end voice signal but also many acoustic coupling components and line wrap-around components. Because it is impossible to estimate the call state accurately by simply comparing the power of the transmitted signal and the received signal, an algorithm for reducing the effects of the acoustic coupling component and the line wraparound component is required. It becomes.

  As a loudspeaker device equipped with a voice switch considering use under a high ambient noise level as described above and application to a system with a large acoustic coupling gain and line wraparound feedback gain, one described in Patent Document 1 is disclosed. is there.

  FIG. 5 is a block diagram showing a loudspeaker call system including the loudspeaker device (master device M) of Patent Document 1 and a call terminal (slave device S) connected to master device M via a two-wire transmission line. FIG. 6 is a block diagram showing a detailed configuration of the voice switch VS mounted on the master unit M. The base unit M includes a microphone 1, a speaker 2, a two-wire / four-wire conversion hybrid circuit 3, a microphone amplifier G2 that amplifies a transmission signal from the microphone 1, and a line output provided on the output side of the transmission-side attenuator 4. It comprises an amplifier G1, a line input amplifier G3 for amplifying a received signal from the line, a speaker amplifier G4, and a voice switch VS. The slave unit S includes a microphone 1 ', a speaker 2', a two-wire / four-wire conversion hybrid circuit 3 ', a microphone amplifier G2', and a speaker amplifier G4 '.

The voice switch VS includes a transmission side attenuator 4 inserted on a transmission signal line for transmitting a voice signal (transmission signal) collected by the microphone 1 to the line, and a voice signal (reception signal) received from the line. ) On the reception signal line for transmitting to the speaker 2, and an insertion loss amount control unit for controlling the gains of the transmission side attenuator 4 and the reception side attenuator 5 according to the call state. 6. Further, the insertion loss amount control unit 6 includes a first instantaneous power estimation unit 7 that estimates an instantaneous power of an input signal (point B signal) to the transmission side attenuator 4 and an input to the reception side attenuator 5. A second instantaneous power estimator 8 for estimating the instantaneous power of the signal (the signal at point C) and the input point B to the transmission side attenuator 4 and the reception side after passing through the transmission side attenuator 4 and the line side. Line feedback gain multiplication means 9 having a value determined according to the gain of the system that feeds back to the input point C to the side attenuator 5 as a coefficient, and the reception side attenuator 5 from the input point C to the reception side attenuator 5 In addition, an acoustic coupling gain multiplication means 10 having a value determined according to the gain of the path reaching the input point B to the transmission side attenuator 4 after passing through the acoustic side, and a second instantaneous power estimation output signal P _D 'and the first instantaneous obtained an output signal P _C part 8 to enter into the acoustic coupling gain multiplying means 10 A first comparator 11 for comparing the magnitude of the output signal P _B of the power estimator 7, obtained by the output signal P _B of the first instantaneous power estimator 7 inputs to the line feedback gain multiplication means 9 The second comparator 12 that compares the magnitude relationship between the output signal P _A ′ and the output signal P _{C of} the second instantaneous power estimator 8, and the output signals of the first comparator 11 and the second comparator 12. And an insertion loss amount distribution processing unit 13 that determines a call state based on C1 and C2 and controls the gains of the transmission side attenuator 4 and the reception side attenuator 5.

Here, the first and second instantaneous power estimation units 7 and 8 are realized by an envelope detector, an integration circuit, or the like having characteristics that the rise is steep and the fall is gradual. And instantaneous powers P _B and P _C of the input signal to the receiving side attenuator 5 are estimated.

  The line feedback gain multiplication means 9 includes a variable coefficient multiplier 9a having a value Gt equal to the gain of the transmission side attenuator 4 as a coefficient and an output point of the transmission side attenuator 4 measured in advance on the line side. A fixed coefficient multiplier 9b having as a coefficient a value ηt obtained by multiplying the gain of the path reaching the input point C of the reception side attenuator 5 through a wraparound by a predetermined margin value (about 2 to 3 times). Further, the acoustic coupling gain multiplication means 10 includes a variable coefficient multiplier 10a having a coefficient Gr equal to the gain of the reception side attenuator 5, and a speaker amplifier G4-speaker from the output point of the reception side attenuator 5 measured in advance. 2-A value obtained by multiplying the gain of the path from the microphone 1 to the input point B of the transmitting-side attenuator 5 through the microphone 1 through the acoustic transmission system and the microphone 1 through a predetermined margin (about 2 to 3 times). a fixed coefficient multiplier 10b having ηr as a coefficient. Here, when setting the coefficients ηt and ηr of the fixed coefficient multipliers 9b and 10b, the margin values are used because of fluctuations in acoustic coupling gain due to changes in the reflection conditions in front of the speaker 2 and the microphone 1, This is because the fluctuation of the line-side wraparound gain due to the change in impedance when the counterpart telephone terminal (slave unit S) is viewed from the four-wire conversion hybrid circuit 3 is absorbed.

  Next, the operation of the voice switch VS will be described.

The first comparator 11 is obtained by inputting the output signal P _B from the first instantaneous power estimator 7 and the output signal P _C from the second instantaneous power estimator 8 to the acoustic coupling gain multiplier 10. The output signal P _D ′ is compared, and when P _B ≧ P _D ′, the output signal C 1 becomes “1”, and when P _B <P _D ′, the output signal C 1 becomes “0”. Further, the second comparator 12, a first instantaneous power estimator 7 of the output signal P _B output signal obtained by inputting to the line feedback gain multiplication means 9 P _A 'and the second instantaneous power estimator 8 and comparing the output signal _{P C, P a '≧ P} C output signal C2 is "1" in the case of, P _a' output signal C2 when the <P _C becomes "0".

  On the other hand, the insertion loss amount distribution processing unit 13 determines the call state based on the binary signals C1 and C2 output from the first and second comparators 11 and 12, and the transmitting side according to the determination result. The gains of the attenuator 4 and the receiving side attenuator 5 are determined. Here, the determination rule for the call state is a transmission state when C1 = C2 = 1, a reception state when C1 = C2 = 0, and an idle state when C1 ≠ C2. When the determination result is the transmission state, the insertion loss amount distribution processing unit 13 sets the gain of the transmission side attenuator 4 to the maximum value and sets the gain of the reception side attenuator 5 to the minimum value. When the determination result is the reception state, the gain of the transmission side attenuator 4 is set to the minimum value, the gain of the reception side attenuator 5 is set to the maximum value, and the determination result is the idle state. The gains of the transmission side attenuator 4 and the reception side attenuator 5 are set to the same value (the square root value of the total gain).

Thus, according to the voice switch VS, by appropriately setting the coefficients ηt and ηr of the fixed coefficient multipliers 9b and 10b, it is possible to reliably block the transmission of speech even in a voice communication system having a large acoustic coupling gain and line feedback gain. Reception blocking can be prevented.
Japanese Patent No. 3709739

  By the way, in a conventional loudspeaker, a special effect on the voice by performing voice signal processing on the voice signal collected by the microphone, for example, an effect of making the voice color and speed (speaking speed) of the speaker different from the actual voice. May be performed. In such a case, a large delay (for example, several hundred milliseconds to several seconds) occurs at the timing when the sneak component due to line feedback or acoustic coupling is input to the voice switch VS for the time required for voice signal processing on the voice signal. Therefore, an error is likely to occur in the call state estimation process in the voice switch, and as a result, the call voice may be interrupted.

  The present invention has been made in view of the above circumstances, and an object of the present invention is to provide a loudspeaker device that can stabilize the operation of a voice switch and prevent call interruption even in a situation where a voice signal is delayed on a transmission line. Is to provide.

In order to achieve the above object, the invention of claim 1 provides a microphone and a speaker, a reception side signal path for transmitting a reception signal transmitted from the other party's telephone terminal to the speaker, and a transmission collected by the microphone. A voice switch that switches the call state between receiving and transmitting by inserting a loss into the transmitting signal path that transmits the signal and sends it to the other party's call terminal, and a delay with respect to the transmitted signal that is input to the voice switch First voice signal processing means for performing accompanying voice signal processing, and second voice signal processing means for performing voice signal processing with a delay on the received signal input to the voice switch. The transmission side attenuation means inserted on the side signal path, the reception side attenuation means inserted on the reception side signal path, and the gains of the transmission side attenuation means and the reception side attenuation means are controlled according to the call state. The insertion loss amount control unit includes a first instantaneous power estimation unit that estimates an instantaneous power of an input signal to the transmission side attenuation unit, and an input signal to the reception side attenuation unit. A second instantaneous power estimator for estimating the instantaneous power of the receiver, and a system that feeds back from the input point to the transmission side attenuation means to the input point to the reception side attenuation means via a wraparound on the transmission side attenuation means and the line side Line feedback gain multiplying means having a value determined according to the gain of the input, the input point to the receiving side attenuating means from the input point to the receiving side attenuating means, and the input point to the transmitting side attenuating means via the wraparound on the acoustic side An acoustic coupling gain multiplying means having a value determined according to the gain of the path to the path, and an output signal obtained by inputting the output signal of the second instantaneous power estimating section to the acoustic coupling gain multiplying means; The magnitude relationship with the output signal of the instantaneous power estimation unit 1 The magnitude comparison between the output signal obtained by inputting the output signal of the first comparator to be compared with the output signal of the first instantaneous power estimation unit to the line coupling gain multiplication means and the output signal of the second instantaneous power estimation unit is compared. A second comparator that determines the call state based on the output signals of the first comparator and the second comparator, and controls the gain of the transmission side attenuation means and the reception side attenuation means. A line feedback gain multiplying means comprising: a variable coefficient multiplier having a coefficient substantially equal to the gain of the transmitting side attenuating means; and receiving a signal from the output point of the transmitting side attenuating means via a loop on the line side. A fixed coefficient multiplier whose coefficient is a value obtained by multiplying the gain of the path to the input point of the side attenuation means by a predetermined margin value, and the acoustic coupling gain multiplication means is substantially equal to the gain of the reception side attenuation means From the variable coefficient multiplier with coefficients and the output point of the receiving side attenuation means A fixed coefficient multiplier whose coefficient is a value obtained by multiplying the gain of the path to the input point of the transmission side attenuation means via the acoustic coupling system by a predetermined margin value, and a group delay in the line feedback gain Second delay means for delaying the input signal by a time corresponding to the group delay in the acoustic coupling gain is provided in the first stage or the subsequent stage of the line feedback gain multiplication means. Is provided before or after the acoustic coupling gain multiplication means.

  According to the first aspect of the present invention, the first delay means for delaying the input signal by a time corresponding to the group delay in the line feedback gain is provided before or after the line feedback gain multiplication means, and the group delay in the acoustic coupling gain is provided. Since the second delay means for delaying the input signal by the time corresponding to is provided at the front stage or the rear stage of the acoustic coupling gain multiplication means, for example, there is a delay (group delay) on the transmission path from the transmission signal inputted from the microphone. Even in the situation that occurs, the first delay means cancels the relative time difference between the received signal and the transmitted signal by delaying the received signal received from the counterpart telephone terminal by the time corresponding to the delay (group delay). As a result, it is possible to stabilize the operation of the voice switch and prevent the call from being interrupted even in a situation where the voice signal is delayed on the transmission line.

According to a second aspect of the present invention, in the first aspect of the invention, the first and second audio signal processing means can selectively switch the presence or absence of the audio signal processing, and the first or second delay means At least one of them is characterized in that the input signal is delayed only when the first or second audio signal processing means performs audio signal processing.

  According to the invention of claim 2, it is possible to prevent malfunction of the voice switch due to unnecessary delay by delaying by the delay means only when there is a delay in the voice signal.

In order to achieve the above object , the invention according to claim 3 is a microphone and a speaker, a reception side signal path for transmitting a reception signal transmitted from the other party's call terminal to the speaker, and a transmission collected by the microphone. A voice switch for switching a call state between receiving and transmitting by inserting a loss in a transmitting side signal path for transmitting a signal and sending it to the other party's telephone terminal; and at least one of a front stage or a rear stage of the voice switch; An echo canceller having an adaptive filter, and the voice switch includes a transmission-side attenuation unit inserted on the transmission-side signal path, a reception-side attenuation unit inserted on the reception-side signal path, and a call state according to a call state And an insertion loss amount control unit for controlling the gain of the transmission side attenuation unit and the reception side attenuation unit. The insertion loss amount control unit is an instantaneous power of an input signal to the transmission side attenuation unit. A first instantaneous power estimator for estimating; a second instantaneous power estimator for estimating an instantaneous power of an input signal to the receiving side attenuating means; a transmitting side attenuating means from an input point to the transmitting side attenuating means; Line feedback gain multiplication means having a coefficient determined by the gain of the system that returns to the input point to the receiving side attenuation means through wraparound on the line side, and the input side to the receiving side attenuation means to the receiving side An acoustic coupling gain multiplying unit having a coefficient determined by a gain determined in accordance with the gain of the path reaching the input point to the transmitting side attenuation unit through the attenuation unit and the sound side attenuation unit; and a second instantaneous power estimation unit The first comparator for comparing the magnitude relationship between the output signal obtained by inputting the output signal to the acoustic coupling gain multiplication means and the output signal of the first instantaneous power estimation unit, and the first instantaneous power estimation unit Obtained by inputting the output signal to the line coupling gain multiplier. The second comparator for comparing the magnitude relationship between the output signal to be output and the output signal of the second instantaneous power estimation unit, and the call state is determined based on the output signals of the first comparator and the second comparator And an insertion loss distribution processor for controlling the gains of the transmission side attenuation means and the reception side attenuation means, and the line feedback gain multiplication means is a variable coefficient multiplication having a coefficient substantially equal to the gain of the transmission side attenuation means. A fixed coefficient multiplier having as a coefficient a value obtained by multiplying the gain of the path from the output point of the transmission side attenuation means to the input point of the reception side attenuation means through the loop side by a predetermined margin value The acoustic coupling gain multiplication means includes a variable coefficient multiplier having a coefficient substantially equal to the gain of the reception side attenuation means, and an input point of the transmission side attenuation means from the output point of the reception side attenuation means through the acoustic coupling system. The coefficient is a value obtained by multiplying the gain of the route to the destination by a predetermined margin value. And a first delay means for delaying the input signal by a time corresponding to the group delay in the line feedback gain, provided at the front stage or the rear stage of the line feedback gain multiplication means, and the acoustic coupling gain. The second delay means for delaying the input signal by a time corresponding to the group delay in the first stage or the rear stage of the acoustic coupling gain multiplication means is provided, and the first and second delay means are filter coefficients of an adaptive filter included in the echo canceller. The delay time is adjusted according to the above.

According to the invention of claim 3, in addition to the operational effect of the invention of claim 1, the first and second delay means adjust the delay time according to the filter coefficient of the adaptive filter of the echo canceller. It is possible to accurately estimate the delay time and reliably stabilize the operation of the voice switch, and it is not necessary to obtain the delay time corresponding to the group delay in advance.

  According to the present invention, for example, even in a situation in which a transmission signal input from a microphone has a delay (group delay) on the transmission path, the first delay unit causes the other party to receive a signal corresponding to the delay (group delay). By delaying the received signal received from the call terminal, the relative time difference between the received signal and the transmitted signal can be canceled out. As a result, the operation of the voice switch is stable even in the situation where the voice signal is delayed on the transmission path. It is possible to prevent interruption of the call.

(Embodiment 1)
Hereinafter, embodiments of the present invention will be described in detail with reference to the drawings. However, since the basic configuration of the loudspeaker device according to the present embodiment is the same as that of the conventional example (master unit M) described in Patent Document 1, the same components are denoted by the same reference numerals and appropriately illustrated and illustrated. Description is omitted.

  FIG. 1 is a block diagram showing a main part of the loudspeaker apparatus (master M) of the present embodiment. In the present embodiment, as shown in FIG. 1, the first audio signal processing unit 50 that performs audio signal processing with a delay on the transmission signal input to the audio switch VS, and the reception signal input to the audio switch VS A second audio signal processing unit 51 that performs audio signal processing with a delay, a first delay unit 60 provided after the line feedback gain multiplication unit 9, and a stage subsequent to the acoustic coupling gain multiplication unit 10. The second delay means 61 is provided.

  The first and second audio signal processing units 50 and 51 perform an audio frequency conversion process for converting an audio frequency (voice color), an audio speed conversion process for converting an audio speed, and the like (not shown). The presence or absence of the audio signal processing can be alternatively switched according to the operation input received by the operation input receiving means. For example, if the first audio signal processing unit 50 performs the audio frequency conversion process, the relatively high voice of a woman or a child is converted into a relatively low voice like a man, so that the sales to the push sale etc. You can respond with peace of mind. Alternatively, when the speech speed conversion process is performed by the second audio signal processing unit 51, it is possible to cause the speaker 2 to sound at a slow speed suitable for hearing impaired persons, elderly people, and the like. However, since any audio signal processing of the audio frequency conversion processing and the speech speed conversion processing can be realized by a conventionally well-known technique (for example, TDHS <Time Domain Harmonic Scaling> algorithm in the speech speed conversion processing). Description is omitted.

  Here, in the first and second audio signal processing units 50 and 51, in order to perform audio signal processing with a delay such as audio frequency conversion processing and speech speed conversion processing on the audio signal once stored in the buffer, A large delay (for example, several hundred milliseconds to several seconds) occurs at the timing when the sneak component due to line feedback or acoustic coupling is input to the voice switch VS depending on whether or not the voice signal processing is performed. When such a large delay occurs, the call state estimation process in the voice switch VS may become unstable and may cause a malfunction.

  Therefore, in the present embodiment, the first delay means 60 is provided after the line feedback gain multiplication means 9 of the voice switch VS, and the second delay means 61 is provided after the acoustic coupling gain multiplication means 10, When the audio signal processing unit 50 is performing audio signal processing, the first delay means 60 delays the received signal input to the audio switch VS by a predetermined delay time (first delay time), and the second audio When the signal processing unit 51 is performing voice signal processing, the second delay means 61 delays the transmission signal input to the voice switch VS by a predetermined delay time (second delay time). However, the first delay time is set to a time corresponding to the group delay in the acoustic coupling gain (a delay time generated by the audio signal processing of the first audio signal processing unit 50), and the second delay time is set to the line feedback gain. It is set to a time corresponding to the group delay (a delay time caused by the audio signal processing of the second audio signal processing unit 51).

Thus, when the first audio signal processing unit 50 performs the audio signal processing, the input of the sneak component due to acoustic coupling to the audio switch VS is delayed, but the line feedback gain is the time corresponding to the delay time. By delaying the timing at which the output signal P _A ′ of the multiplication means 9 is input to the second comparator 12 by the first delay means 60, the delay due to the audio signal processing of the first audio signal processing section 50 is first. Therefore, the time difference between the input timings of the two signals P _A ′ and P _C compared by the second comparator 12 is eliminated. Similarly, when the second audio signal processing unit 51 performs audio signal processing, the input of the sneak component due to line feedback to the audio switch VS is delayed, but the acoustic coupling gain multiplication is performed for a time corresponding to the delay time. By delaying the timing at which the output signal P _D ′ of the means 10 is input to the first comparator 11 by the second delay means 61, the delay due to the audio signal processing of the second audio signal processing section 51 is reduced to the second. The time difference between the input timings of the two signals P _D ′ and P _B which are canceled by the delay by the delay means 61 and compared by the first comparator 11 is eliminated. Therefore, even when the first and second voice signal processing units 50 and 51 perform voice signal processing, it is difficult for an error to occur in the call state estimation process of the voice switch VS, and the voice of the call due to blocking of the voice switch VS is reduced. Interruption can be prevented. However, when the first and second audio signal processing units 50 and 51 do not perform audio signal processing, the first and second delay means 60 and 61 do not delay the audio signal, and the audio is caused by unnecessary delay. It is desirable to prevent malfunction of the switch VS. In the present embodiment, the first and second delay means 60 and 61 are provided in the subsequent stage of the line feedback gain multiplication means 9 and the acoustic coupling gain multiplication means 10, respectively, but are provided in the previous stage of the means 9 and 10. It doesn't matter.

(Embodiment 2)
In the present embodiment, as shown in FIG. 2, an acoustic echo canceller (not shown) that suppresses an acoustic echo generated by acoustic coupling between the microphone 1 and the speaker 2 and an acoustic coupling or signal in a transmission system at the other party's call terminal A line-side echo canceller 30 that suppresses line echo caused by wraparound and a delay time estimation unit 62 that estimates a delay time according to the filter coefficient of the adaptive filter 31 included in the line-side echo canceller 30. A voice switch VS is arranged between the side echo cancellers 30. However, since the basic configuration of the present embodiment is the same as that of the first embodiment, illustration and description of the common components are appropriately omitted.

As shown in FIG. 3, the line-side echo canceller 30 includes reflection due to impedance mismatch between the 2-wire / 4-wire conversion hybrid circuit 3 and the transmission path, and between the speaker 2 ′ and the microphone 1 ′ in the doorphone slave unit S. An adaptive filter 31 that adaptively identifies an impulse response of a feedback path (line echo path) H _LIN formed by acoustic coupling, and an echo component (line echo) estimated from a reference signal (transmission signal) from the received signal And a subtractor 32 for subtracting. The adaptive filter 31 updates the filter coefficient based on a predetermined algorithm (for example, a learning identification method or the like). An acoustic echo canceller (not shown) has the same configuration as the line echo canceller 30.

Here, the relationship between the tap number and the filter coefficient (count value) in the adaptive filter 31 is shown in FIG. 4, and the D taps in FIG. 4 correspond to the group delay τ _T in the echo path. When the sampling period is Ts, the following relationship is established.

τ _T = D · Ts
Therefore, the delay time estimation unit 62 calculates the group delay τ _T based on the above equation when the line-side echo canceller 30 is in the converged state, and uses the group delay τ _T as the delay time in the first delay means 60. Set.

  As described above, in the present embodiment, the delay time of the first delay means 60 is adjusted according to the filter coefficient of the adaptive filter 31 included in the line-side echo canceller 30. Therefore, the delay time is accurately estimated and the voice switch VS is used. Can be reliably stabilized, and it is not necessary to obtain the delay time corresponding to the group delay in advance. However, in the present embodiment, only the line-side echo canceller (line-side echo canceller 30) has been described, but the acoustic-side echo canceller (acoustic-side echo canceller) is configured in the same manner, and the adaptive filter of the acoustic-side echo canceller is used. It is possible to adjust the delay time of the second delay means 61 according to the filter coefficient.

It is a block diagram which shows Embodiment 1 of this invention. It is a block diagram of the principal part which shows Embodiment 2 of this invention. It is a block diagram which shows the line side echo canceller 30 in the same as the above. It is operation | movement explanatory drawing of the delay time estimation part in the same as the above. It is a block diagram which shows a prior art example. It is a block diagram which shows the voice switch in the same as the above.

Explanation of symbols

M Master unit (loudspeaker)
VS audio switch 50 first audio signal processor 51 second audio signal processor 60 first delay means 61 second delay means

Claims (3)

The microphone and speaker, the receiver side signal path for transmitting the reception signal sent from the other party's telephone terminal to the speaker, and the transmitter side transmitting the transmission signal collected by the microphone to the other party's telephone terminal A voice switch for switching a call state between reception and transmission by inserting a loss in the signal path; first voice signal processing means for performing voice signal processing with a delay on a transmission signal input to the voice switch; A second voice signal processing means for performing voice signal processing with a delay on the received signal input to the voice switch ;
The voice switch includes a transmission side attenuation means inserted on the transmission side signal path, a reception side attenuation means inserted on the reception side signal path, and the transmission side attenuation means and the reception side according to the call state. An insertion loss amount control unit for controlling the gain of the attenuation means,
The insertion loss amount control unit includes a first instantaneous power estimation unit that estimates the instantaneous power of the input signal to the transmission side attenuation unit, and a second instantaneous power that estimates the instantaneous power of the input signal to the reception side attenuation unit. A coefficient that is determined according to the gain of the estimator and the system that feeds back from the input point to the transmission side attenuation means to the transmission side attenuation means and the input point to the reception side attenuation means through wraparound on the line side Line feedback gain multiplying means and the gain of the path from the input point to the receiving side attenuating means to the input side to the receiving side attenuating means and the input side to the transmitting side attenuating means through the wraparound on the acoustic side. And an output signal obtained by inputting the output signal of the second instantaneous power estimator to the acoustic coupling gain multiplier and the output signal of the first instantaneous power estimator A first comparator for comparing magnitude relationships, and a first moment A second comparator for comparing the magnitude relationship between the output signal obtained by inputting the output signal of the power estimation unit to the line coupling gain multiplier and the output signal of the second instantaneous power estimation unit; and the first comparator And an insertion loss amount distribution processing unit that determines a call state based on an output signal of the second comparator and controls a gain of the transmission side attenuation unit and the reception side attenuation unit,
The line feedback gain multiplying means includes a variable coefficient multiplier having a coefficient substantially equal to the gain of the transmitting side attenuating means, and from the output point of the transmitting side attenuating means to the input point of the receiving side attenuating means through wraparound on the line side. A fixed coefficient multiplier whose coefficient is a value obtained by multiplying the gain of the route to be reached by a predetermined margin value;
The acoustic coupling gain multiplication means includes a variable coefficient multiplier having a coefficient substantially equal to the gain of the reception side attenuation means, and a path from the output point of the reception side attenuation means to the input point of the transmission side attenuation means through the acoustic coupling system. A fixed coefficient multiplier whose coefficient is a value obtained by multiplying the gain of a predetermined margin value,
The first delay means for delaying the input signal by a time corresponding to the group delay in the line feedback gain is provided at the front stage or the rear stage of the line feedback gain multiplying means, and the input signal is delayed by the time corresponding to the group delay in the acoustic coupling gain. The loudspeaker apparatus characterized in that the second delay means to be provided is provided in the front stage or the rear stage of the acoustic coupling gain multiplication means. The first and second audio signal processing means can alternatively switch the presence or absence of the audio signal processing, and at least one of the first or second delay means is the first or second audio signal. 2. The loudspeaker apparatus according to claim 1, wherein the input signal is delayed only when the processing means performs voice signal processing. The microphone and speaker, the receiver side signal path for transmitting the reception signal sent from the other party's telephone terminal to the speaker, and the transmitter side transmitting the transmission signal collected by the microphone to the other party's telephone terminal A voice switch that switches the call state to reception and transmission by inserting a loss in the signal path, and an echo canceller that is provided in at least one of the front stage or the rear stage of the voice switch and has an adaptive filter ;
The voice switch includes a transmission side attenuation means inserted on the transmission side signal path, a reception side attenuation means inserted on the reception side signal path, and the transmission side attenuation means and the reception side according to the call state. An insertion loss amount control unit for controlling the gain of the attenuation means,
The insertion loss amount control unit includes a first instantaneous power estimation unit that estimates the instantaneous power of the input signal to the transmission side attenuation unit, and a second instantaneous power that estimates the instantaneous power of the input signal to the reception side attenuation unit. A coefficient that is determined according to the gain of the estimator and the system that feeds back from the input point to the transmission side attenuation means to the transmission side attenuation means and the input point to the reception side attenuation means through wraparound on the line side Line feedback gain multiplying means and the gain of the path from the input point to the receiving side attenuating means to the input side to the receiving side attenuating means and the input side to the transmitting side attenuating means through the wraparound on the acoustic side. And an output signal obtained by inputting the output signal of the second instantaneous power estimator to the acoustic coupling gain multiplier and the output signal of the first instantaneous power estimator A first comparator for comparing magnitude relationships, and a first moment A second comparator for comparing the magnitude relationship between the output signal obtained by inputting the output signal of the power estimation unit to the line coupling gain multiplier and the output signal of the second instantaneous power estimation unit; and the first comparator And an insertion loss amount distribution processing unit that determines a call state based on an output signal of the second comparator and controls a gain of the transmission side attenuation unit and the reception side attenuation unit,
The line feedback gain multiplying means includes a variable coefficient multiplier having a coefficient substantially equal to the gain of the transmitting side attenuating means, and from the output point of the transmitting side attenuating means to the input point of the receiving side attenuating means through wraparound on the line side. A fixed coefficient multiplier whose coefficient is a value obtained by multiplying the gain of the route to be reached by a predetermined margin value;
The acoustic coupling gain multiplication means includes a variable coefficient multiplier having a coefficient substantially equal to the gain of the reception side attenuation means, and a path from the output point of the reception side attenuation means to the input point of the transmission side attenuation means through the acoustic coupling system. A fixed coefficient multiplier whose coefficient is a value obtained by multiplying the gain of a predetermined margin value,
The first delay means for delaying the input signal by a time corresponding to the group delay in the line feedback gain is provided at the front stage or the rear stage of the line feedback gain multiplying means, and the input signal is delayed by the time corresponding to the group delay in the acoustic coupling gain. A second delay means to be provided in the front stage or the rear stage of the acoustic coupling gain multiplication means,
The first and second delay means adjust the delay time according to the filter coefficient of the adaptive filter included in the echo canceller.

Images