JP2005347875A - Loudspeaking speech apparatus - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a loudspeaking speech apparatus capable of making a speech even at a place such as a bath room with many background noises and a large reverberation. <P>SOLUTION: When a background noise level included in a transmission signal exceeds a threshold value, an operating mode of a voice switch 10 is set to a fixed mode, the sensitivity of the voice switch 10 is corrected and a sound volume of a speaker 2 is increased. As a result, even when a level of various noises produced in the bath room exceeds a prescribed threshold value, since a total loss amount calculation section 14 operated in the fixed mode fixes a total loss to a sufficiently greater initial value, production of unpleasant echo and howling can be suppressed, the voice switch 10 is easily switched from a transmission state to a reception state so as to prevent sole selection of the transmission state due to a high level noise in the bath room, since the sound volume of the speaker 2 is increased, a pleasant speech can be realized. <P>COPYRIGHT: (C)2006,JPO&NCIPI

Description

  The present invention relates to a loudspeaker device used for intercoms and the like, and more particularly to a loudspeaker device suitable for installation in a bathroom.

  In this type of loudspeaker, an uncomfortable echo (acoustic echo or acoustic echo) is caused by an acoustic return path formed by acoustic coupling of a microphone and a speaker, or a line-side return path formed between the other party's call terminal. Line echo) may be heard, or howling occurs at the frequency when a closed loop in which the loop gain in an arbitrary frequency component exceeds 1 is formed in the communication system by the feedback path or the like. In some cases, an echo canceller and a voice switch are provided to prevent the generation of unpleasant echoes and howling as described above.

  The voice switch constantly estimates the call state (sending state, receiving state) and inserts a loss into the signal path on the transmitting side and the receiving side with an appropriate distribution based on the estimation result. The echo canceller adaptively identifies the impulse response of the feedback path and estimates the pseudo echo component of the feedback path from the input signal to the feedback path, and the pseudo echo component estimated by the adaptive filter as the feedback path. And a subtracter that subtracts from the output signal from the. Here, the adaptive filter of the echo canceller usually requires several seconds of learning time to identify the impulse response of the feedback path, so the echo suppression effect by the echo canceller cannot be expected sufficiently in the few seconds immediately after the start of the call. There is a possibility that an unpleasant echo or howling may occur because a closed loop is formed in the call system.

  Therefore, the present applicant has already proposed a loudspeaker device that can suppress unpleasant echoes and howling immediately after the start of a call (see Patent Document 1).

In this conventional example, when the echo canceller immediately after the start of the call has not converged, the voice switch operates in a fixed mode that fixes the total amount of loss (total loss amount) inserted into the signal path to a sufficiently large initial value. Thus, unpleasant echoes and howling are suppressed, and in a state where the echo canceller has sufficiently converged, the voice switch operates in an update mode in which the total loss amount is updated at any time, thereby realizing two-way simultaneous calls.
JP 2002-359580 A

  By the way, in a general intercom system, a loudspeaker is often installed in a living room of a house, but recently it may be installed in a bathroom. In the bathroom, there are many background noises such as the sound of hot water discharged from a currant (mixed faucet) and shower head and the exhaust sound of a ventilating fan, and the echo is larger than in the living room. It could be difficult.

  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 capable of making a call even in a place such as a bathroom where there is a lot of background noise and a large echo.

  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. Suppresses acoustic echo caused by acoustic coupling between the microphone and speaker, and a voice switch that switches the call state between receiving and transmitting by inserting loss into the transmitting signal path that transmits the signal and sends it to the other party's telephone terminal An echo canceller, and a voice switch includes transmission side loss insertion means for inserting loss into the signal path on the transmission side, reception side loss insertion means for inserting loss into the signal path on the reception side, transmission side and Insertion loss amount control means for controlling the amount of loss inserted from each loss insertion means on the receiver side. -Estimate the acoustic feedback gain of the path that returns to the input point of the transmission side loss insertion means via the path, and the reception side loss insertion means from the output point of the transmission side loss insertion means via the line echo path. A total loss amount calculation unit that estimates the line-side feedback gain of the path that returns to the input point and calculates the sum of the loss amounts to be inserted into the closed loop based on the estimated feedback gain values on the acoustic side and the line side; The conversation state is estimated by monitoring the speech signal and the reception signal, and the distribution of the insertion loss amounts of the transmission side loss insertion means and the reception side insertion loss means is allocated according to the estimation result and the calculated value of the total loss amount calculation unit. An insertion loss amount distribution processing unit to be determined, and the total loss amount calculation unit calculates the sum of loss amounts to be inserted into the closed loop based on the estimated value of each feedback gain and adaptively updates the update mode, and the total loss Fixed mode that fixes the amount to a predetermined initial value Operating in fixed mode during the period from the start of communication with the other party's call terminal until the echo canceller converges sufficiently, and during the period after the echo canceller converges sufficiently In a loudspeaker device operating in the update mode, background noise level detecting means for detecting a background noise level included in a transmission signal collected by a microphone, and background noise level detected by the background noise level detecting means A background noise level determining means for outputting a control command when the threshold value is exceeded when compared with a predetermined threshold; a volume adjusting means for increasing the volume of the received signal when receiving the control command from the background noise level determining means; When a control command is received from the noise level judgment means, the total loss amount calculation unit of the voice switch is set to the fixed mode and the insertion loss amount is distributed. The present invention is characterized by comprising mode setting means for making it easier to estimate the call state estimation processing in the processing unit as the reception state.

  According to a second aspect of the present invention, in the loudspeaker device according to the first aspect, instead of the background noise level detection means and the background noise level determination means, a shower flow rate detection means for detecting a flow rate of a shower installed in the bathroom; A shower flow rate determination unit is provided that compares a shower flow rate detected by the shower flow rate detection unit with a predetermined threshold value and outputs a control command when the threshold value is exceeded.

  A third aspect of the present invention is the loudspeaker apparatus according to the first aspect, wherein the mixed faucet flow rate for detecting the flow rate of the mixed faucet installed in the bathroom instead of the background noise level detecting means and the background noise level determining means. It comprises a detecting means and a mixed faucet flow determining means for comparing the mixed faucet flow detected by the mixed faucet flow detecting means with a predetermined threshold and outputting a control command when the threshold is exceeded. To do.

  According to a fourth aspect of the present invention, in the loudspeaker apparatus according to the first aspect, instead of the background noise level detecting means and the background noise level determining means, the air volume of a heating facility for heating and heating the bathroom is detected. An air volume detecting means and an air volume determining means for comparing the air volume detected by the air volume detecting means with a predetermined threshold value and outputting a control command when the threshold value is exceeded are provided.

  According to a fifth aspect of the present invention, in the loudspeaker apparatus according to the first aspect, in place of the background noise level detecting means and the background noise level judging means, the air volume of the drying equipment that sends hot air into the bathroom and simultaneously ventilates the bathroom. And an air volume determination means for comparing the air volume detected by the air volume detection means with a predetermined threshold value and outputting a control command when the threshold value is exceeded.

  According to a sixth aspect of the present invention, in the loudspeaker apparatus according to the first aspect, in place of the background noise level detecting means and the background noise level determining means, an air volume detecting means for detecting an air volume of a ventilation facility for ventilating the inside of the bathroom, and an air volume An air volume determining means for comparing the air volume detected by the detecting means with a predetermined threshold value and outputting a control command when the threshold value is exceeded is provided.

  A seventh aspect of the present invention is the loudspeaker communication apparatus according to the first aspect, wherein instead of the background noise level detection means and the background noise level determination means, the suction of the jet bath equipment that sucks in the hot water in the bathtub and jets the sucked hot water A water amount detecting means for detecting at least one of the amount and the ejection amount, and a water amount determining means for comparing the suction amount or the ejection amount detected by the water amount detecting means with a predetermined threshold value and outputting a control command when the threshold value is exceeded It is characterized by comprising.

  According to the present invention, even when the level of various noises generated in the bathroom exceeds a predetermined threshold, the total loss amount is fixed to a sufficiently large initial value by the total loss amount calculation unit operating in the fixed mode. Therefore, the occurrence of unpleasant echo and howling is suppressed, and the voice switch is changed from the transmission state to the reception state by making it easier to estimate the call state estimation processing in the insertion loss distribution processing unit as the reception state. This makes it easy to switch and prevents the phone from falling down to the transmission state. Further, since the volume of the speaker is increased by the volume adjusting means, a comfortable call can be realized.

  Hereinafter, embodiments in which the present invention is applied to a loudspeaker communication apparatus (interphone terminal) installed in a bathroom of a house will be described in detail with reference to the drawings.

(Embodiment 1)
In the present embodiment, as shown in FIG. 1, a microphone 1, a speaker 2, a two-wire / four-wire conversion circuit 3, a microphone amplifier G1, and a line output amplifier G2 that amplifies a transmission signal to a line (two-wire transmission line). A line input amplifier G3 for amplifying a reception signal from the line, a speaker amplifier G4, a transmission volume adjustment amplifier G5, a reception volume adjustment amplifier G6, a voice switch 10, and first and second echo cancellers 30A and 30B. ing.

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 The acoustic echo is adaptively identified by the adaptive filter 31A and subtracted from the output signal of the microphone amplifier G1 by the subtractor 32A from the pseudo echo component (acoustic echo) estimated from the reference signal (input signal to the speaker amplifier G4). It is to suppress. The second echo canceller 30B also has a conventionally well-known configuration including an adaptive filter 31B and a subtractor 32B. The reflection due to impedance mismatch between the 2-wire-to-wire conversion circuit 3 and the transmission path and the counterpart The impulse response of the feedback path (line echo path) H _LIN formed by the acoustic coupling between the speaker and the microphone in the telephone terminal (for example, intercom system door phone slave unit) is adaptively identified by the adaptive filter 31B and referenced. The line echo is suppressed by subtracting the pseudo echo component (line echo) estimated from the signal (input signal to the line output amplifier G2, that is, transmission signal) from the reception signal by the subtractor 32B.

The voice switch 10 includes a transmission side attenuator 11 for inserting a loss in the signal path on the transmission side, a reception side attenuator 12 for inserting a loss in the signal path on the reception side, and attenuations on the transmission side and the reception side. And an insertion loss amount control unit 13 for controlling the loss amount inserted from the devices 11 and 12. The insertion loss amount control unit 13, the path to be fed back from the output point Rout of the receiving side attenuator 12 to the input point Tin of the transmitter-side attenuator 11 via the acoustic echo path H _AC (hereinafter referred to as "acoustic side feedback path" ) On the acoustic side feedback gain α and a path for returning from the output point Tout of the transmitting side attenuator 11 to the input point Rin of the receiving side attenuator 12 via the line echo path H _LIN (hereinafter referred to as “line side”). The line-side feedback gain β of the feedback path ”is estimated, and the total amount of loss to be inserted into the closed loop based on the estimated values α ′ and β ′ of the feedback gains α and β on the acoustic side and the line side (transmission) The total loss amount calculation unit 14 for calculating the insertion loss amount of the side attenuator 11 and the insertion loss amount of the reception side attenuator 12, and the call state is estimated by monitoring the transmission signal and the reception signal. The transmission side attenuator 11 and the reception side decrease according to the result and the calculated value of the total loss calculation unit 14 It comprises an insertion loss amount distribution processing unit 15 that determines the distribution of each insertion loss amount of the attenuator 12. The first and second echo cancellers 30A and 30B and the voice switch 10 in this embodiment are controlled by controlling the DSP (Digital Signal Processor) hardware with software (program) for the echo canceller and the voice switch. It has been realized. 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 quantized by the A / D converter. It has become.

The total loss amount calculation unit 14 estimates the time-average power of the input signal of the transmission side attenuator 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 attenuator 12 using a time average power of the output signal of the maximum delay receiving side attenuator 12 at the time envisaged by the acoustic side feedback path H _AC A minimum value of the estimated value is obtained, and a value obtained by dividing the estimated value of the time average power of the input signal of the transmission side attenuator 11 by this minimum value is used as an estimated value α ′ of the acoustic feedback gain α, and a rectifier / smoothing device The time average power of the input signal of the reception side attenuator 12 is estimated in a short time using a low pass filter or the like, and the output signal of the transmission side attenuator 11 is similarly used using a rectifier smoother or a low pass filter. In a short time Estimating the average power, determining the minimum value of the estimated value of the time average power of the output signal of the transmitter-side attenuator 11 at the maximum delay time assumed in the line side feedback path H _LIN, receiving side attenuation at this minimum value A value obtained by dividing the estimated value of the time average power of the input signal of the unit 12 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 transmitting side attenuator 11 and the input / output signals of the receiving side attenuator 12, and information such as the magnitude relationship between the power levels of these signals and the presence / absence of an audio signal. The communication state (the reception state, the transmission state, etc.) is determined from the transmission state, and the total loss Lt is distributed to the transmission side attenuator 11 and the reception side attenuator 12 at a rate corresponding to the determined call state. The amount of insertion loss of the attenuators 11 and 12 is adjusted.

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

  By the way, since the reverberation becomes large in a relatively narrow space such as a bathroom, it takes a long time for the echo canceller to converge as compared with a relatively large living space (low reverberation space). Here, in the adaptive filter constituted by the digital FIR filter, the filter coefficient is sequentially corrected by an algorithm that operates so as to minimize the erasure error that is not erased by subtraction of the pseudo echo component. In the conventional example described, an algorithm (for example, LMS (Least-Mean-Square) method) that minimizes the root mean square value of the erasure error is used. In this LMS method, a correction range (step gain) for adjusting the correction magnitude of the filter coefficient is given as a scalar quantity, and the convergence time for a colored signal such as an audio signal in a high reverberation space such as a bathroom. Since it becomes considerably long, there is a problem that the time during which the voice switch operates in the fixed mode becomes relatively long immediately after the start of the call, and the state where only one-way call can be made continues for a long time.

  Therefore, in the present embodiment, the adaptive filter 31A included in the first echo canceller 30A is configured by a digital FIR filter, and an algorithm that operates so as to minimize an erasure error that has not been eliminated by subtraction of the pseudo echo component. A step gain matrix represented by a diagonal matrix is used to sequentially correct the filter coefficients and adjust the magnitude of correction of the filter coefficients. In the present embodiment, a conventionally known projection method is used as the above algorithm instead of the LMS method. The projection method improves the convergence speed for a colored signal such as an audio signal by removing the autocorrelation of the input signal inside the algorithm. The filter coefficient (also referred to as tap coefficient) h (n) of the adaptive filter 31A is sequentially corrected according to the following equation (1) by the secondary projection method.

h (n + 1) = h (n) + μ [δ (n) × (n) + ε (n) × (n−1)] (1)
However,
h (n) = (h ₁ (n), h ₂ (n),..., h _L (n)) ^T
T : Vector transposition n: Sampling time L: Tap length (number of taps)
μ: Step gain (scalar amount)
x (n) = (x (n), x (n-1),..., x (n-L + 1)) ^T : input signal (received signal) vector δ (n), ε (n) This is a constant obtained from simultaneous equations (2) and (3).

δ (n) × (n) ^T x (n) + ε (n) × (n−1) ^T x (n) = e (n) (2)
δ (n) × (n−1) ^T x (n) + ε (n) × (n−1) ^T x (n−1) = (1−μ) e (n−1) (3)
However, e (n) is the difference (erasure error) between the true echo component and the pseudo echo component.

  In this embodiment, the filter coefficient h (n) of the adaptive filter 31A is obtained by combining a so-called ES method, which expands the step gain μ given as a scalar quantity into a diagonal matrix called a step gain matrix M, with the projection method. ) Are sequentially corrected according to the following equation (4).

h (n + 1) = h (n) + M [δ (n) x (n) + ε (n) x (n-1)] (4)
However,
M = diag [μ ₁ , μ ₂ ,..., Μ _L ]
μ _i = μ ₀ λ ^i-1 (i = 1, 2,..., L)
λ: Attenuation rate of impulse response fluctuation (0 <λ ≦ 1)
Here, since the output (impulse response) when the impulse is input to the FIR filter becomes the filter coefficient itself, the magnitude of the correction of the filter coefficient coincides with the fluctuation amount of the impulse response in the installation space (for example, bathroom). It will be. In general, it is known that the impulse response in the room is exponentially attenuated regardless of the degree of reverberation, and the fluctuation amount of the impulse response is also attenuated at the same attenuation rate as the impulse response. Therefore, in the ES method, weighting is performed so that the filter coefficient at the initial stage of the impulse response with large fluctuation is corrected with a large step gain, and the filter coefficient at the later stage of the impulse response with small fluctuation is corrected with a small step gain. Specifically, the diagonal elements μ _i (i = 1, 2,..., L) of the step gain matrix M are changed from the maximum value μ ₀ to the attenuation characteristic of the impulse response as i increases as shown in FIG. By attenuating with the same inclination, the convergence time can be shortened as a result.

And Thus, the adaptive filter 31A, an input signal taken at each sampling cycle (received signal) and the received signal vector x (n), x (n ) T x (n), x (n-1) T x ( n), x (n-1) ^T x (n), x (n-1) ^T x (n-1) are calculated, and the erase error e (n) stored in the memory and the step gain matrix M are paired. The constants δ (n) and ε (n) are obtained by reading the angular element μ _i and solving the simultaneous equations of the equations (2) and (3), and the obtained constants δ (n) and ε (n) and the memory. The second term on the right-hand side of the equation (4) is calculated using the step gain matrix M read out from, and added to the filter coefficient h (n) read out from the memory to obtain the next filter coefficient h (n + 1). By performing the calculation, the filter coefficient h (n + 1) is sequentially corrected, and a process of bringing the filter coefficient h (n + 1) closer to the true impulse response is performed.

  As described above, according to the present embodiment, the filter coefficients are adaptively identified by the algorithm of the ES projection method that combines the projection method and the ES method in the adaptive filter 31A of the first echo canceller 30A. Compared with the LMS method or the learning identification method, the convergence time of the filter coefficient in a high reverberation space such as a bathroom can be shortened. In addition, in the present embodiment, until the first echo canceler 30A converges, the voice switch 10 is operated in the fixed mode to realize a half-duplex call that suppresses the occurrence of unpleasant echoes and howling, and the first switch When the echo canceller 30A converges, the voice switch 10 is operated in the update mode to realize two-way simultaneous communication. By shortening the convergence time of the first echo canceller 30A, the voice switch 10 operates in the fixed mode. It is possible to shorten the period during which the call is made, that is, the period during which the half-duplex call is made, and to make an early transition to a two-way simultaneous call. As a result, a comfortable voice call can be made even in a highly reverberant space such as a bathroom.

By the way, there is provided step gain matrix setting means for reading out the diagonal elements μ _i (i = 1, 2,..., L) of the step gain matrix M stored in the memory in advance and setting them in the adaptive filter 31A. If it is a structure, a value with a high approximation accuracy to an impulse response can be used as a diagonal element, and there exists an advantage that the convergence time of the 1st echo canceller 30A can be shortened reliably. Note that the step gain matrix setting means is realized by controlling the DSP hardware by software as in the echo canceller 30A. However, as the tap length L increases, the number of diagonal elements μ _i also increases, which may increase the memory storage area and increase the cost.

Therefore, as shown in FIG. 4, the step gain matrix setting for setting the diagonal elements μ _i of the step gain matrix M by approximating the diagonal elements μ _i by the three parameters of the maximum value μ ₀ , the attenuation factor λ, and the setting interval D. If the means is provided, the storage area of the memory can be greatly reduced and the diagonal element μ _i can be easily adjusted. However, the configuration in which the diagonal element μ _i is set by three parameters as described above has a demerit that the approximation accuracy to the impulse response is lower than the above configuration in which each diagonal element μ _i is stored in the memory. The step gain matrix setting means reads out each diagonal element μ _i from the memory and creates the step gain matrix M, and the diagonal element with the three parameters of the maximum value μ ₀ , the attenuation factor λ, and the set interval D The process of creating the step gain matrix M by approximating μ _i in a staircase pattern may be alternatively switched and executed. Then, the setting method of the diagonal element μ _i of the step gain matrix M can be changed to an appropriate method according to conditions such as the magnitude of reverberation at the installation location.

  By the way, there are many background noises such as the sound of hot water discharged from a currant or shower head or the exhaust sound of a ventilation fan in the bathroom, and the space is narrower and the response is larger than in the living room. Calls may be difficult due to noise. For example, when the voice switch 10 is operated in the update mode when the noise (background noise) generated when hot water is discharged from a currant or shower head is large, the total loss amount inserted in the closed loop is compared with the background noise level. There is a possibility that a phenomenon (so-called “one-sided fall”) in which the voice switch 10 is fixed in the transmission mode occurs and the voice of the visitor cannot be heard.

  Therefore, in the present embodiment, the background noise level detection unit 40 that detects the background noise level included in the transmission signal collected by the microphone 1 and the background noise level detected by the background noise level detection unit 40 are set to a predetermined threshold value. The background noise level determination unit 41 that outputs a control command when the threshold is exceeded, and the total loss amount calculation unit 14 of the voice switch 10 is set to the fixed mode when the control command is received from the background noise level determination unit 41. A mode setting / volume adjustment unit that increases the volume of the speaker 2 by setting and increasing the volume of the speaker amplifier G4 by making it easier to estimate the call state estimation process in the insertion loss distribution processing unit 15 as the reception state and adjusting the gain of the speaker amplifier G4 42. However, the background noise level detection unit 40, the background noise level determination unit 41, and the mode setting / volume adjustment unit 42 in the present embodiment are realized by controlling DSP hardware by software.

  The background noise level detection unit 40 is composed of a digital filter having a characteristic that the rise is gradual and the fall is steep, that is, a response characteristic having a relatively large rise time constant and a relatively small fall time constant, This is to detect (estimate) the background noise level that is constantly present in the transmitted signal. Then, the background noise level determination unit 41 compares the background noise level detected by the background noise level detection unit 40 with a predetermined threshold, and controls the mode setting / volume adjustment unit 42 when the background noise level exceeds the threshold. Outputs a command. Note that the threshold value to be compared with the background noise level may be set to the maximum level at which a person can talk under the background noise level, for example.

  On the other hand, when the mode setting / volume adjusting unit 42 receives the control command from the background noise level determining unit 41, the mode setting / volume adjusting unit 42 sets the total loss amount calculating unit 14 of the voice switch 10 to the fixed mode and also performs the loss amount distribution process of the voice switch 10. The process for estimating the call state in the unit 15 is relatively easy to estimate as the reception state, and the volume of the speaker 2 is increased by adjusting the gain of the speaker amplifier G4. Here, it is relatively easy to estimate the call state estimation process in the loss amount distribution processing unit 15 as the reception state, in other words, the sensitivity of the voice switch 10 (ease of switching between the transmission state and the reception state) is the reception state. As a specific method for making the signal level higher when switching from the transmission state to the reception state than when switching from the state to the transmission state, for example, multiplying the signal level of the reception signal by a coefficient greater than 1, Alternatively, the received signal is weighted by multiplying the signal level of the transmitted signal by a coefficient less than 1, and the signal level of the received signal exceeds the signal level of the transmitted signal in comparison between the signal level of the transmitted signal and the received signal. There are methods to make it easier.

  Thus, when another interphone terminal is called from the loudspeaker device of the present embodiment, before the call with the counterpart terminal is started, or when called from another interphone terminal, the loudspeaker device of the present embodiment Before the call is started after the response button (not shown) is operated, the operation mode of the voice switch 10 is set to the fixed mode and the sensitivity of the voice switch 10 is corrected as described above. If the level of various noises generated in the bathroom exceeds a predetermined threshold value, the total loss amount is fixed to a sufficiently large initial value by the total loss amount calculation unit 14 operating in the fixed mode. Therefore, the generation of unpleasant echoes and howling is suppressed, and the voice switch 10 is easily switched from the transmitting state to the receiving state, and the noise in the bathroom is increased. Falling pieces to transmission state is prevented by further since the volume of the speaker 2 is increased, in which comfortable call can be realized.

(Embodiment 2)
As shown in FIG. 5, the present embodiment replaces the background noise level detection unit 40 and the background noise level determination unit 41 with a shower flow rate detection unit 43 that detects the flow rate of a shower installed in the bathroom, and the bathroom. The flow rate detection unit 44 for detecting the flow rate of the installed curan, the shower flow rate detection unit 43 and the shower flow rate and the current flow rate detected by the current flow rate detection unit 44 are respectively compared with a predetermined threshold, and at least one of them There is a feature in that it includes a shower / curran flow rate determination unit 45 that outputs a control command to the mode setting / volume adjustment unit 42 when the flow rate exceeds a corresponding threshold value. However, since the configuration other than this is the same as that of the first embodiment, the same reference numerals are assigned to the common components, and illustration and description are omitted as appropriate.

  The shower flow rate detection unit 43 and the curan flow rate detection unit 44 include a flow rate sensor installed in a pipe from the mixing tap that mixes the hot water supplied from the water heater and the water supplied from the water supply pipe to the shower head and the current flow, And a signal processing circuit for outputting a flow rate detection signal obtained by performing signal processing (amplification, waveform shaping, etc.) on the sensor output of the sensor. Note that the flow rate sensor is a conventionally known one such as an impeller type, an electromagnetic induction type, or an ultrasonic type.

  Thus, when another interphone terminal is called from the loudspeaker device of the present embodiment, before the call with the counterpart terminal is started, or when called from another interphone terminal, the loudspeaker device of the present embodiment The flow rate detection signal output from the shower flow rate detection unit 43 and the flow rate detection signal output from the currant flow rate detection unit 44 before the start of a call after the response button (not shown) is operated at When it is input to the flow rate determination unit 45 and the shower / curan flow rate determination unit 45 compares the detected flow rate of the shower and the detected flow rate of the current with a predetermined threshold value, and at least one of the detected flow rates exceeds the threshold value A control command is output to mode setting / volume adjustment unit 42. As a result, the operation mode of the voice switch 10 is set to the fixed mode and the sensitivity of the voice switch 10 is corrected and the volume of the speaker 2 is increased as in the first embodiment, so that hot water is discharged from the shower or currant. Even in a situation where the level of noise generated at the time is relatively large, since the total loss amount is fixed to a sufficiently large initial value by the total loss amount calculation unit 14 operating in the fixed mode, unpleasant echoes and howling are generated. And the voice switch 10 is easily switched from the transmitting state to the receiving state, preventing the speaker from falling down to the transmitting state due to loud noise in the bathroom, and further increasing the volume of the speaker 2. Therefore, a comfortable call can be realized. In this embodiment, the shower flow rate detection unit 43 and the currant flow rate detection unit 44 are provided, but a configuration including only one of them may be used.

(Embodiment 3)
By the way, in recent years, it has a circulation fan that circulates the air in the bathroom and a heat exchanger for the bathroom that heats the air by heat exchange with hot water from the water heater, and the bathroom heating ventilation that performs ventilation, heating, and drying of the bathroom Although dryers are widely used, air is blown into the bathroom or during the operation of such a bathroom heating / ventilating dryer, noise is generated according to the air volume when air is blown out from the bathroom, and showers are generated. This is the same noise source as the hot water discharge of KARAN.

  Therefore, in this embodiment, instead of the background noise level detection unit 40 and the background noise level determination unit 41 as shown in FIG. 6, heating equipment that sends warm air into the bathroom for heating, and ventilation equipment that ventilates the bathroom. The air volume detection unit 46 that detects the air volume of the bathroom heating / ventilation dryer that also serves as a drying facility that ventilates the bathroom at the same time as sending warm air into the bathroom, and the air volume detected by the air volume detection unit 46 as a predetermined threshold value An air volume determination unit 47 that outputs a control command to the mode setting / volume adjustment unit 42 when the threshold value is exceeded in comparison.

  Here, the air volume during heating, ventilation, and drying operations of the bathroom heating / ventilation dryer is only switched to several predetermined stages (for example, “strong” and “weak”). In the embodiment, the air volume detection unit 46 monitors the control signal for air volume adjustment output from the remote controller of the bathroom heating ventilation dryer installed in the dressing room outside the bathroom, and the air volume ( "Strong" or "weak") is detected, and an air volume detection signal representing the detected air volume is output to the air volume determination unit 47.

  Thus, when another interphone terminal is called from the loudspeaker device of the present embodiment, before the call with the counterpart terminal is started, or when called from another interphone terminal, the loudspeaker device of the present embodiment Before a call is started after a response button (not shown) is operated, an air volume detection signal output from the air volume detection unit 46 is input to the air volume determination unit 47, and the air volume determination unit 47 performs bathroom heating ventilation dryer. The detected air volume is compared with a predetermined threshold value, and when the detected air volume exceeds the threshold value, a control command is output to the mode setting / volume adjusting unit 42. As a result, as in the first and second embodiments, the operation mode of the voice switch 10 is set to the fixed mode, the sensitivity of the voice switch 10 is corrected, and the volume of the speaker 2 is increased. Even in a situation where the level of noise generated according to the air volume during operation is relatively large, the total loss amount is fixed to a sufficiently large initial value by the total loss amount calculation unit 14 operating in the fixed mode, so that an unpleasant echo And howling is suppressed, and the voice switch 10 is easily switched from the transmission state to the reception state, so that it is prevented from falling down to the transmission state due to loud noise in the bathroom. Therefore, a comfortable call can be realized.

(Embodiment 4)
By the way, recently, even in ordinary houses, a circulation water channel that connects the suction portion and the ejection portion of the bathtub is provided in communication, and is operated by a motor in the middle of the circulation water channel, and hot water in the bathtub is sucked from the suction portion and the bathtub from the ejection portion. Jet bath equipment with a pump unit that jets out is being installed, but hot water in the bathtub is sucked in or water is jetted into the bathtub during operation of such a jet bath equipment. Noise is generated according to the amount of suction or ejection (hereinafter referred to as “water volume”), and it is the same noise source as hot water discharge from showers and currants and hot air blowout from bathroom heating ventilation dryers. Yes.

  Therefore, in this embodiment, in place of the background noise level detection unit 40 and the background noise level determination unit 41 as shown in FIG. 7, the suction amount and the ejection of the jet bath facility that sucks in the hot water in the bathtub and ejects the sucked hot water. A water amount detection unit 48 that detects at least one of the amounts, and a mode setting / volume adjustment unit 42 that compares the suction amount or the ejection amount detected by the water amount detection unit 48 with a predetermined threshold value and exceeds a threshold value. And a water amount determination unit 49 for outputting a control command.

  Here, since the suction amount or the ejection amount of the jet bath facility can only be switched to a predetermined number of stages (for example, two stages of “strong” and “weak”), in this embodiment, in the bathroom The control signal for adjusting the suction amount or the ejection amount output from the remote controller of the installed jet bath facility is monitored by the water amount detection unit 48, and the suction amount or the ejection amount (“strong” or “ Weak ") is detected, and a detection signal indicating the detected suction amount or ejection amount is output to the water amount determination unit 49.

  Thus, when another interphone terminal is called from the loudspeaker device of the present embodiment, before the call with the counterpart terminal is started, or when called from another interphone terminal, the loudspeaker device of the present embodiment Before a call is started after a response button (not shown) is operated, a water amount detection signal output from the water amount detection unit 48 is input to the water amount determination unit 49, and the water amount determination unit 49 detects the jet bath facility. The water amount is compared with a predetermined threshold value, and a control command is output to the mode setting / volume adjusting unit 42 when the detected water amount exceeds the threshold value. As a result, as in the first to third embodiments, the operation mode of the voice switch 10 is set to the fixed mode, the sensitivity of the voice switch 10 is corrected, and the volume of the speaker 2 is increased. Even in a situation where the level of noise generated according to the amount of water (suction amount or ejection amount) is relatively high, the total loss amount is fixed to a sufficiently large initial value by the total loss amount calculation unit 14 operating in the fixed mode. Therefore, the occurrence of unpleasant echo and howling is suppressed, the voice switch 10 is easily switched from the transmission state to the reception state, and the falling to the transmission state due to the loud noise in the bathroom is prevented, Furthermore, since the volume of the speaker 2 is increased, a comfortable call can be realized.

It is a block diagram which shows Embodiment 1 of this invention. It is a flowchart for operation | movement description of a voice switch in the same as the above. It is operation | movement explanatory drawing of the 1st echo canceller in the same as the above. It is operation | movement explanatory drawing of the 1st echo canceller in the same as the above. It is a block diagram which shows the principal part of Embodiment 2 of this invention. It is a block diagram which shows the principal part of Embodiment 3 of this invention. It is a block diagram which shows the principal part of Embodiment 4 of this invention.

Explanation of symbols

DESCRIPTION OF SYMBOLS 1 Microphone 2 Speaker 10 Voice switch 30A 1st echo canceller 31A Adaptive filter 32A Subtractor 40 Background noise level detection part 41 Background noise level determination part 42 Mode setting / volume adjustment part

Claims (7)

  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 The voice switch includes a voice switch that switches a call state between reception and transmission by inserting a loss in the signal path, and an echo canceller that suppresses an acoustic echo generated by acoustic coupling between a microphone and a speaker. Controls the amount of loss inserted from the transmission side loss insertion means for inserting loss into the path, the reception side loss insertion means for inserting loss into the signal path on the reception side, and the loss insertion means on the transmission side and reception side. Insertion loss amount control means, the insertion loss amount control means from the output point of the reception side loss insertion means to the input point of the transmission side loss insertion means via the acoustic echo path Estimate the acoustic side feedback gain of the return path, and estimate the line side feedback gain of the path returning from the output point of the transmission side loss insertion means to the input point of the reception side loss insertion means via the line echo path, A total loss amount calculation unit that calculates the total amount of loss to be inserted into the closed loop based on the estimated values of the feedback gains on the acoustic side and the line side, and estimates the call state by monitoring the transmission signal and the reception signal, The total loss amount is composed of an insertion loss amount distribution processing unit that determines the distribution of each insertion loss amount of the transmission side loss insertion means and the reception side insertion loss means according to the estimation result and the calculated value of the total loss amount calculation unit. The calculation unit calculates two sums of loss amounts to be inserted into the closed loop based on the estimated values of the feedback gains, an update mode for adaptively updating, and a fixed mode for fixing the total loss amount to a predetermined initial value. Mode, the other party's call terminal In a loudspeaker that operates in the fixed mode during the period from when the call starts until the echo canceller converges sufficiently, and in the update mode during the period after the echo canceller converges sufficiently, A background noise level detecting means for detecting a background noise level included in the transmitted transmission signal, a background noise level detected by the background noise level detecting means is compared with a predetermined threshold value, and a control command is issued when the threshold value is exceeded. The output of the background noise level judging means, the volume adjusting means for increasing the volume of the received signal when receiving the control command from the background noise level judging means, and the voice switch of the voice switch when receiving the control command from the background noise level judging means The total loss amount calculation unit is set to the fixed mode and the call state estimation processing in the insertion loss amount distribution processing unit is relatively received. A loudspeaker having a mode setting means for facilitating estimation of a talk state.   2. The loudspeaker apparatus according to claim 1, wherein instead of the background noise level detecting means and the background noise level determining means, a shower flow rate detecting means for detecting a flow rate of a shower installed in the bathroom, and a shower flow rate detecting means. And a shower flow rate judging means for outputting a control command when the shower flow rate is compared with a predetermined threshold value and exceeds the predetermined threshold value.   2. The loudspeaker apparatus according to claim 1, wherein instead of the background noise level detecting means and the background noise level determining means, a mixed faucet flow rate detecting means for detecting a flow rate of the mixed faucet installed in the bathroom, and a mixed faucet A loudspeaker communication device comprising: a mixed faucet flow rate judging means for comparing the mixed faucet flow rate detected by the flow rate detecting means with a predetermined threshold value and outputting a control command when the threshold value is exceeded.   2. The loudspeaker apparatus according to claim 1, wherein instead of the background noise level detection means and the background noise level determination means, an air volume detection means for detecting the air volume of a heating facility for sending warm air into the bathroom and heating, and an air volume detection A loudspeaker apparatus comprising: an air volume determining unit that compares an air volume detected by the means with a predetermined threshold value and outputs a control command when the threshold value is exceeded.   The loudspeaker apparatus according to claim 1, wherein instead of the background noise level detecting means and the background noise level determining means, an air volume detecting means for detecting the air volume of a drying facility that sends hot air into the bathroom and simultaneously ventilates the bathroom. A loudspeaker apparatus comprising: an air volume determination unit that compares an air volume detected by the air volume detection unit with a predetermined threshold value and outputs a control command when the threshold value is exceeded.   2. The loudspeaker according to claim 1, wherein instead of the background noise level detecting means and the background noise level determining means, an air volume detecting means for detecting an air volume of a ventilation facility for ventilating the inside of the bathroom, and an air volume detected by the air volume detecting means. A loudspeaker apparatus comprising: an air volume determination unit that compares a predetermined threshold value and outputs a control command when the threshold value is exceeded.   2. The loudspeaker apparatus according to claim 1, wherein at least one of a suction amount and a jet amount of a jet bath facility that sucks hot water in the bathtub and jets the sucked hot water in place of the background noise level detection means and the background noise level determination means. A water amount detecting means for detecting either of the above and a water amount determining means for comparing a suction amount or an ejection amount detected by the water amount detecting means with a predetermined threshold value and outputting a control command when the threshold value is exceeded. A voice call device.

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