JP4458002B2 - Loudspeaker - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a voice amplifying phone for carrying out voice amplification with a proper voice volume in response to the magnitude of surrounding noise and suppressing production of howling at the same time. <P>SOLUTION: Since an estimate value Pn' of a near end side surrounding noise level is updated when a near end side voice interval detection section detects no voice interval, a voice volume correction amount adjustment means 6 adjusts a correction amount in a voice volume correction means 4 to a proper value in response to a surround noise level so as to attain a voice amplifying speech in a proper voice volume in response to the magnitude of the surrounding noise. Since the voice volume correction amount adjustment means 6 adjusts the correction amount only when the voice interval is detected, production of howling caused by a sneak path from a speaker 2 to a microphone 1 can be suppressed. Further, since a voice volume correction means 40 adopts a received signal level adjustment means provided in a voice switch VS, the cost reduction by simplifying the configuration can be attained in comparison with separate provision of the voice volume correction means 40. <P>COPYRIGHT: (C)2007,JPO&amp;INPIT

Description

  The present invention relates to a loudspeaker communication apparatus such as an interphone that includes a microphone and a speaker and performs a loudspeaker call.

In a conventional communication device, for example, an interphone master unit equipped with a handset, it is transmitted from the doorphone child unit via a communication line when talking with a doorphone child unit that is a loudspeaker device equipped with a microphone and a speaker instead of the handset. The ambient noise level (far-end side ambient noise level) contained in the received signal is estimated, and the received signal and the level of the transmitted signal transmitted to the intercom unit via the telephone line are adjusted based on the estimated value. By doing so, the voice of the visitor can be heard at an appropriate volume (for example, see Patent Document 1).
JP 2002-185625 A

  However, in the interphone master unit configured as a loudspeaker device that uses a microphone and a speaker instead of a handset as in the case of the doorphone slave unit, the wraparound component to the microphone is also increased by increasing the speaker volume. Therefore, howling is likely to occur under circumstances where ambient noise is high. In addition, when a voice switch that switches between the transmission state and the reception state is provided to prevent howling, a phenomenon (receiving blocking) that makes it difficult for the voice switch to switch to the reception state occurs due to an increase in the sneak component from the speaker to the microphone. There was a problem that.

  The present invention has been made in view of the above circumstances, and an object of the present invention is to make a voice call with an appropriate volume corresponding to the level of ambient noise and at the same time suppress the occurrence of howling and blocking. To provide an apparatus.

In order to achieve the above-mentioned object, the invention according to claim 1 provides a microphone, a speaker, a transmission state in which a transmission signal output from the microphone is transmitted to the far end side, and a reception signal transmitted from the far end side in the speaker. A voice switch for selectively switching the reception state to be input to, a near-end side ambient noise level estimation means for estimating a near-end side ambient noise level included in the transmission signal, and increasing / decreasing the received signal level A sound volume correcting means for correcting the sound volume of the sound produced by the speaker, a far-end-side sound section detecting means for detecting a sound section in which the received signal includes a sound component, and a far-end-side sound section detecting means is a sound section. A volume correction amount adjusting means for adjusting a correction amount in the volume correction means in accordance with the ambient noise level estimated by the near-end side ambient noise level estimation means when detecting Receiving signal level adjusting means for increasing / decreasing the received signal level, the received signal level adjusting means serving as the volume correcting means, and the near-end side ambient noise level estimating means for a short time of the instantaneous power of the transmitted signal A short-time average value calculating unit that calculates an average value and a long-time average value calculating unit that calculates a long-time average value of the instantaneous power, and comparing the short-time average value with a long-time average value A near-end side speech section detecting unit that detects a speech section in which the transmitted signal includes a speech component; and an ambient noise level calculating unit that calculates an estimated value of the near-end side ambient noise level included in the transmitted signal. And when the near-end side speech section detection unit is not detecting a speech section and updates the estimated value of the near-end side ambient noise level and the near-end side speech section detection unit is detecting a speech section Near-end ambient noise level A loudspeaker telephone device which does not update the estimated value, volume correction amount adjusting means, while the voice switch is switched to the receiving state, characterized in that not adjusted correction amount.

In order to achieve the above object , the invention according to claim 2 provides a microphone, a speaker, a transmission state in which a transmission signal output from the microphone is transmitted to the far end side, and a reception signal transmitted from the far end side in the speaker. A voice switch for selectively switching the reception state to be input to, a near-end side ambient noise level estimation means for estimating a near-end side ambient noise level included in the transmission signal, and increasing / decreasing the received signal level A sound volume correcting means for correcting the sound volume of the sound produced by the speaker, a far-end-side sound section detecting means for detecting a sound section in which the received signal includes a sound component, and a far-end-side sound section detecting means is a sound section. A volume correction amount adjusting means for adjusting a correction amount in the volume correction means in accordance with the ambient noise level estimated by the near-end side ambient noise level estimation means when detecting Receiving signal level adjusting means for increasing / decreasing the received signal level, the received signal level adjusting means serving as the volume correcting means, and the near-end side ambient noise level estimating means for a short time of the instantaneous power of the transmitted signal A short-time average value calculating unit that calculates an average value and a long-time average value calculating unit that calculates a long-time average value of the instantaneous power, and comparing the short-time average value with a long-time average value A near-end side speech section detecting unit that detects a speech section in which the transmitted signal includes a speech component; and an ambient noise level calculating unit that calculates an estimated value of the near-end side ambient noise level included in the transmitted signal. And when the near-end side speech section detection unit is not detecting a speech section and updates the estimated value of the near-end side ambient noise level and the near-end side speech section detection unit is detecting a speech section Near-end ambient noise level A Not updating estimates hands-free communication device, a volume correction amount adjusting means, while the voice switch is switched to the receiving state and performs the adjustment of the correction amount after a predetermined delay time.

According to a third aspect of the present invention, in the first or second aspect of the invention, the long-time average value calculation unit does not update the long-time average value while the voice switch is switched to the receiving state.

According to a fourth aspect of the present invention, in the first or second aspect of the present invention, the voice switch refers to the detection result of the far-end side voice section detecting means and the detection result of the near-end side voice section detecting unit to determine the call state. It is characterized by switching.

According to a fifth aspect of the present invention, in the first or second aspect of the present invention, the voice switch includes transmission side loss insertion means for inserting loss into the signal path of the transmission signal, and reception for inserting loss into the signal path of the reception signal. Side loss insertion means and insertion loss amount control means for controlling the amount of loss inserted from each of the transmission side and reception side loss insertion means, and the insertion loss amount control means is an input to the transmission side loss insertion means. A first instantaneous power estimator for estimating the instantaneous power of the signal; a second instantaneous power estimator for estimating the instantaneous power of the input signal to the receiving-side loss insertion means; and an input point to the transmitting-side loss insertion means. From the transmission side loss insertion means and the far end side feedback gain multiplication means having as a coefficient a value determined in accordance with the gain of the system that returns to the input point to the reception side loss insertion means through wraparound at the far end side , The loss on the receiving side from the input point to the loss insertion means on the receiving side A near-end side combined gain multiplier having a coefficient as a value determined in accordance with a gain of a path reaching the input point to the transmission-side loss insertion means through the insertion means and the wraparound at the near-end side; A first comparator for comparing a magnitude relationship between an output signal obtained by inputting an output signal of the instantaneous power estimation unit to the near-end side combined gain multiplication means and an output signal of the first instantaneous power estimation unit; A second comparator for comparing the magnitude relationship between the output signal obtained by inputting the output signal of the instantaneous power estimator to the far-end feedback gain multiplier and the output signal of the second instantaneous power estimator; An insertion loss amount distribution processing unit that determines a call state based on output signals of the first comparator and the second comparator and controls gains of the transmission side loss insertion unit and the reception side loss insertion unit; The far-end side feedback gain multiplication means is approximately the same as the gain of the transmission side loss insertion means. A variable coefficient multiplier having a new coefficient and the gain of the path from the output point of the transmission side loss insertion means to the input point of the reception side loss insertion means via the wraparound on the far end side is multiplied by a predetermined margin value. A fixed coefficient multiplier having a value as a coefficient, and the near-end side combined gain multiplication means includes a variable coefficient multiplier having a coefficient substantially equal to the gain of the reception side loss insertion means, and an output point of the reception side loss insertion means A fixed coefficient multiplier whose coefficient is a value obtained by multiplying the gain of the path from the speaker through the speaker and microphone to the input point of the transmission side loss insertion means by a predetermined margin value, and estimating the near-end side ambient noise level The coefficient of the variable coefficient multiplier is increased as the ambient noise level estimated by the means increases.

According to a sixth aspect of the present invention, in the first or second aspect of the present invention, the voice switch includes transmission side loss insertion means for inserting loss into the signal path of the transmission signal, and reception for inserting loss into the signal path of the reception signal. Side loss insertion means and insertion loss amount control means for controlling the amount of loss inserted from each of the transmission side and reception side loss insertion means, and the insertion loss amount control means is an input to the transmission side loss insertion means. A first instantaneous power estimator for estimating the instantaneous power of the signal; a second instantaneous power estimator for estimating the instantaneous power of the input signal to the receiving-side loss insertion means; and an input point to the transmitting-side loss insertion means. From the transmission side loss insertion means and the far end side feedback gain multiplication means having as a coefficient a value determined in accordance with the gain of the system that returns to the input point to the reception side loss insertion means through wraparound at the far end side , The loss on the receiving side from the input point to the loss insertion means on the receiving side A near-end side combined gain multiplier having a coefficient as a value determined in accordance with a gain of a path reaching the input point to the transmission-side loss insertion means through the insertion means and the wraparound at the near-end side; A first comparator for comparing a magnitude relationship between an output signal obtained by inputting an output signal of the instantaneous power estimation unit to the near-end side combined gain multiplication means and an output signal of the first instantaneous power estimation unit; A second comparator for comparing the magnitude relationship between the output signal obtained by inputting the output signal of the instantaneous power estimator to the far-end feedback gain multiplier and the output signal of the second instantaneous power estimator; An insertion loss amount distribution processing unit that determines a call state based on output signals of the first comparator and the second comparator and controls gains of the transmission side loss insertion unit and the reception side loss insertion unit; The near-end side coupling gain multiplication means is substantially the same as the gain of the receiving side loss insertion means. A variable coefficient multiplier having a new coefficient and a gain obtained by multiplying the gain of the path from the output point of the receiving side loss insertion means through the speaker and microphone to the input point of the transmission side loss insertion means by a predetermined margin value. The far-end side feedback gain multiplier means includes a variable coefficient multiplier having a coefficient substantially equal to the gain of the transmission side loss insertion means, and an output point of the transmission side loss insertion means. A fixed coefficient multiplier having as a coefficient a value obtained by multiplying the gain of the path that reaches the input point of the receiving-side loss insertion means through the wrapping at the far-end side, and the volume correction amount adjusting means In the case where the volume is increased by increasing the correction amount, the coefficient of the fixed coefficient multiplier is decreased.

A seventh aspect of the invention is characterized in that, in the first or second aspect of the invention, the speaker is a plane wave speaker having a structure for vibrating a flat plate-shaped vibrating body.

The invention of claim 8 is characterized in that, in the invention of claim 1 or 2 , the microphone is a microphone having directivity.

The invention of claim 9 is characterized in that, in the invention of claim 1 or 2 , a housing in which a microphone and a speaker are disposed on the front side is provided, and the speaker is disposed vertically above the microphone on the front surface of the housing. To do.

The invention of claim 10 is the invention of claim 9 , wherein a plurality of types of operation buttons including a call button for instructing the start of a call are arranged between a speaker and a microphone on the front surface of the housing. And

The invention of claim 11 is characterized in that, in the invention of claim 9 , the microphone is a plurality of directional microphones arranged in parallel in the horizontal direction.

According to the first aspect of the present invention, the near-end side ambient noise level estimation means updates the near-end side ambient noise level estimate when the near-end side speech section detector does not detect a speech section and Since the estimated value of the near-end side ambient noise level is not updated when the end-side speech section detection unit detects the speech section, the correction amount in the volume correction unit is appropriately adjusted by the volume correction amount adjustment unit according to the ambient noise level. The volume correction amount adjusting means is adjusted only when the loudspeaking call can be performed at an appropriate volume according to the level of the ambient noise, and the far end side voice section detecting means detects the voice section. Since the amount of correction in the volume correction unit is adjusted according to the ambient noise level estimated by the near-end side ambient noise level estimation unit, the volume from the speaker to the microphone is reduced by not performing volume correction in the non-voice section. It is possible to suppress the occurrence of the cause was howling and blocking. In addition, since the received signal level adjusting means provided in the voice switch is used as the volume correction means, the cost can be reduced by simplifying the configuration as compared with the case where the volume correction means is provided separately. Furthermore, it is possible to prevent the volume of the sound of the sound of the speaker from being raised or lowered during the call, thereby improving the call quality.

According to the invention of claim 2 , the near-end side ambient noise level estimation means updates the near-end side ambient noise level estimate when the near-end side speech section detector does not detect a speech section and Since the estimated value of the near-end side ambient noise level is not updated when the end-side speech section detection unit detects the speech section, the correction amount in the volume correction unit is appropriately adjusted by the volume correction amount adjustment unit according to the ambient noise level. The volume correction amount adjusting means is adjusted only when the loudspeaking call can be performed at an appropriate volume according to the level of the ambient noise, and the far end side voice section detecting means detects the voice section. Since the amount of correction in the volume correction unit is adjusted according to the ambient noise level estimated by the near-end side ambient noise level estimation unit, the volume from the speaker to the microphone is reduced by not performing volume correction in the non-voice section. It is possible to suppress the occurrence of the cause was howling and blocking. In addition, since the received signal level adjusting means provided in the voice switch is used as the volume correction means, the cost can be reduced by simplifying the configuration as compared with the case where the volume correction means is provided separately. Furthermore, even under a situation where the ambient noise level on the near end side frequently fluctuates, it is possible to prevent the sound volume heard from the speaker from increasing or decreasing in the middle of a call, thereby preventing a reduction in call quality.

According to the invention of claim 3 , the estimated value of the ambient noise level by the near-end side ambient noise level estimation means does not change, and as a result, the occurrence of howling and blocking can be prevented.

According to the invention of claim 4 , the call state switching accuracy is improved.

According to the invention of claim 5 , if the coefficient of the variable coefficient multiplier of the near-end side combined gain multiplication means is increased as the ambient noise level increases, the voice switch can be easily switched to the reception state, and the ambient noise level is reduced. The occurrence of reception blocking under relatively high conditions can be prevented.

According to the invention of claim 6 , if the coefficient of the fixed coefficient multiplier of the far-end side feedback gain multiplication means is decreased as the volume increases, it becomes difficult for the voice switch to switch to the transmission state, and from the speaker to the microphone. It is possible to prevent the reception blocking from occurring when the volume increases under the condition that the wraparound is large.

According to the seventh aspect of the present invention, it is difficult for the sound produced by the speaker to be collected by the microphone, and the occurrence of howling due to the acoustic coupling between the speaker and the microphone can be suppressed, thereby improving the call quality.

According to the eighth aspect of the present invention, it is difficult for the sound produced by the speaker to be collected by the microphone, and howling due to the acoustic coupling between the speaker and the microphone can be suppressed, thereby improving the call quality.

According to the ninth aspect of the present invention, the positional relationship between the speaker's ear and the speaker and the positional relationship between the speaker's mouth and the microphone are optimized, and the occurrence of howling due to the acoustic coupling between the speaker and the microphone can be suppressed. At the same time, the call voice can be transmitted to the speaker's ear at an optimum volume.

According to the invention of claim 10 , there is no dead space on the front surface of the housing.

According to the eleventh aspect of the present invention, it is possible to collect noise equivalent to the ambient noise reaching the speaker's ear and correct the sound volume to an optimum value.

  Below, two-way loudspeaker calls (hands-free calls) are made with the dwelling units (interphone master unit, housing information panel, etc.) installed at the common entrance (lobby) of the apartment building and installed in each unit of the apartment building. An embodiment in which the technical idea of the present invention is applied to a lobby intercom that performs the above will be described. However, the loudspeaker device to which the present invention is applicable is not limited to the lobby intercom, and for example, the technical idea of the present invention can be applied to a dwelling unit installed in each dwelling unit.

(Embodiment 1)
FIG. 1 shows a block diagram of Embodiment 1 of the present invention. In the present embodiment, the microphone 1 and the speaker 2, a transmission state in which a transmission signal output from the microphone 1 is transmitted to the far end side, and a reception state in which a reception signal transmitted from the far end side is input to the speaker 2. A voice switch VS for selectively switching, a near-end side ambient noise level estimation means 30 for estimating a near-end side ambient noise level included in a transmission signal output from the microphone 1, and a reception input to the speaker 2 Volume correction means 40 for correcting the volume of the sound that the speaker 2 rings by increasing / decreasing the signal level, a far-end side voice section detection means 50 for detecting a voice section in which the received signal includes a voice component, and a far-end The correction amount in the sound volume correction unit 40 is determined according to the ambient noise level estimated by the near-end side ambient noise level estimation unit 30 when the side speech segment detection unit 50 detects a speech segment. And a sound volume correcting amount adjusting means 60 for settling. In the present embodiment, each of the near end side ambient noise level estimating means 30, the sound volume correcting means 40, the far end side voice section detecting means 50, the sound volume correction amount adjusting means 60, and the sound switch VS are connected to a DSP (digital). Realized by controlling hardware such as a signal processor and CPU with dedicated software. Therefore, the voice signal (received signal) transmitted from the other party's call device and the voice signal (transmitted signal) output from the microphone 1 are quantized to a digital value by an A / D converter (not shown) and are sent to the speaker 2. An input voice signal (received signal) and a voice signal (transmitted signal) transmitted to the other party's communication device are converted into analog values by a D / A converter (not shown).

The voice switch VS includes a transmission side loss insertion unit 4 for inserting a loss into a signal path of a transmission signal (hereinafter referred to as “transmission path”), and a signal path of the reception signal (hereinafter referred to as “reception path”). A receiving-side loss insertion unit 5 for inserting a loss into the transmission side loss insertion unit 5 and an insertion loss amount control unit 6 for controlling the insertion loss amount of the transmission-side loss insertion unit 4 and the reception-side loss insertion unit 5 according to the call state; A transmission signal amplifier G ₂ for amplifying a transmission signal provided in the transmission path on the microphone 1 side from the transmission side loss insertion section 4, and a reception path on the far end side from the reception side loss insertion section 5. And a transmission signal amplifier (volume correction means) 40 for amplifying the received signal. Also, the insertion loss amount control unit 6 is connected to the first instantaneous power estimation unit 7 that estimates the instantaneous power of the input signal (point B signal) to the transmission side loss insertion unit 4 and the reception side loss insertion unit 5. A second instantaneous power estimator 8 that estimates the instantaneous power of the input signal (signal at point C), the input point B to the transmitter side loss inserter 4 and the transmitter side loss inserter 4 and the far end side. The far-end side 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 reception-side loss insertion unit 5 through the wraparound and the reception-side loss insertion unit 5 The near end having a coefficient determined by the gain of the path from the input point C to the input point B to the transmission side loss insertion unit 4 after passing through the reception side loss insertion unit 5 and the near end side. the side coupling gain multiplying means 10, input an output signal P _C of the second instantaneous power estimator 8 to the proximal end side coupling gain multiplying means 10 A first comparator 11 for comparing the output signal P _D 'obtained the magnitude relationship between the output signal P _B of the first instantaneous power estimator 7, the output signal P of the first instantaneous power estimator 7 _A second comparator 12 that compares the magnitude relationship between the output signal P _A ′ obtained by inputting _B to the far-end feedback gain multiplier 9 and the output signal P _{C of} the second instantaneous power estimation unit 8; Insertion that determines the call state based on the output signals C1 and C2 of the first comparator 11 and the second comparator 12 and controls the insertion loss amount of the transmission side loss insertion unit 4 and the reception side loss insertion unit 5 And a loss amount distribution processing unit 13.

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. The instantaneous powers P _B and P _C of the input signal to the receiver and the input signal to the receiver-side loss insertion unit 5 are estimated.

Further, the far-end side feedback gain multiplication means 9 includes a variable coefficient multiplier 9a having a coefficient Gt equal to the gain of the transmission side loss insertion unit 4 and an output point of the transmission side loss insertion unit 4 measured in advance. A fixed coefficient multiplier 9b having a coefficient ηt obtained by multiplying the gain of the path reaching the input point C of the receiving-side loss insertion unit 5 through the wraparound at the far end side by a predetermined (about 2 to 3 times) margin value. And have. Further, the near-end side combined gain multiplying means 10 includes a variable coefficient multiplier 10a having a coefficient Gr equal to the gain of the reception side loss insertion unit 5 and a speaker from the output point of the reception side loss insertion unit 5 measured in advance. 2-A predetermined margin (about 2 to 3 times) in the gain of the path from the microphone 1 to the input point B of the transmission side loss insertion section 4 through the transmission signal amplifier G ₂ and the acoustic transmission system to the microphone 1 And a fixed coefficient multiplier 10b having a coefficient 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 variations in the near-end side coupling gain due to changes in the reflection conditions in front of the speaker 2 and the microphone 1, This is to absorb the variation in the far-end feedback gain due to the change in impedance when the side communication device is viewed.

  The voice switch VS configured as described above operates as follows.

In the first comparator 11, 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 are input to the near-end side combined gain multiplier 10. The obtained output signal P _D ′ is compared. When P _B ≧ P _D ′, the output signal C1 becomes “1”, and when P _B <P _D ′, the output signal C1 becomes “0”. . In the second comparator 12, the output signal P _A 'obtained by inputting the output signal P _B of the first instantaneous power estimation unit 7 to the far-end feedback gain multiplication means 9 and the second instantaneous power estimation. Compared with the output signal P _C of the unit 8, the output signal C2 becomes “1” when P _A ′ ≧ P _C , and the output signal C2 becomes “0” when P _A ′ <P _C. .

  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 insertion loss amounts of the loss insertion unit 4 and the reception side loss insertion unit 5 are determined. Here, the determination rule of 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 a transmission state, the insertion loss amount distribution processing unit 13 minimizes the insertion loss amount of the transmission side loss insertion unit 4 and the insertion loss amount of the reception side loss insertion unit 5. Is set to the maximum value, and conversely, when the determination result is the reception state, the insertion loss amount of the transmission side loss insertion unit 4 is set to the maximum value, and the insertion loss amount of the reception side loss insertion unit 5 is set to the minimum value. Further, when the determination result is the idle state, the insertion loss amounts of the transmission side loss insertion unit 4 and the reception side loss insertion unit 5 are set to equal values.

  As shown in FIG. 2, the near-end side ambient noise level estimation means 30 includes a short-time average value calculation unit 31 that calculates a short-time average value Ps of instantaneous power of an input signal (transmission signal), and a long-time instantaneous power. The long-term average value calculating unit 32 that calculates the average value Pn, and the near-end side voice that detects the voice section in which the transmission signal includes the voice component by comparing the short-time average value Ps and the long-time average value Pn. A section detection unit 33 and an ambient noise level calculation unit 34 that calculates an estimated value Pn ′ of the near-end side ambient noise level included in the transmission signal.

  The short-time average value calculation unit 31 multiplies the instantaneous value (absolute value) Pv (n) of the input signal by a positive constant ρ1 (<1) and the delayed short-time average value Ps (n−1). The short-time average value Ps (n) is calculated by performing a process of adding a value obtained by multiplying the value by a positive constant (1-ρ1), that is, an arithmetic process of the following expression (1).

Ps (n) = (1−ρ1) × Ps (n−1) + ρ1 × Pv (n) (1)
The long-time average value calculation unit 32 multiplies the instantaneous value Pv (n) of the input signal by a positive constant ρ2 (0 <ρ2 <ρ1 <1) and the delayed long-time average value Pn (n− The long-time average value Pn (n) is calculated by performing a process of adding a value obtained by multiplying 1) by a positive constant (1-ρ2), that is, an arithmetic process of the following formula (2).

Pn (n) = (1-ρ2) × Pn (n−1) + ρ2 × Pv (n) (2)
The near-end side speech section detection unit 33 compares the ratio (= Ps (n) / Pn (n)) between the short time average value Ps (n) and the long time average value Pn (n) with a predetermined threshold δ. , If δ <Ps (n) / Pn (n), it is determined as a speech segment if Ps (n) / Pn (n) ≦ δ, and if it is determined as a speech segment, a near-end speech segment is detected. The flag SDF1 is set to 1, and the near-end speech section detection flag SDF1 is set to 0 when it is determined as a non-speech section.

  The ambient noise level calculation unit 34 is a positive constant for the instantaneous value Pv (n) of the input signal when the near-end speech section detection flag SDF1 is 0, that is, when a non-speech section of the transmission signal is detected. A value obtained by multiplying ρ3 (ρ3 <1, where ρ3 may be a different value or the same value as ρ2) and a delayed ambient noise level Pn ′ (n−1) are set to a positive constant (1−ρ3). The ambient noise level Pn ′ (n) is calculated by performing a process of adding the multiplied values, that is, a calculation process of the following expression (3). However, when the near-end side speech section detection flag SDF1 is 1, that is, when the speech section of the transmission signal is detected, the processing of the following expression (3) is not performed and the ambient noise level Pn ′ (n) is set. It is not updated (see the following formula (4)).

Pn ′ (n) = (1−ρ3) × Pn ′ (n−1) + ρ3 × Pv (n) (3)
Pn ′ (n) = Pn ′ (n−1) (4)
Similar to the near-end side ambient noise level estimation unit 30, the far-end side speech section detection unit 50 obtains a short-time average value and a long-time average value of the received signal, and the ratio of both average values is larger than a predetermined threshold value. For example, the far end side speech section detection flag SDF2 is set to 1 when it is determined as a speech section, and the far end side speech section detection flag SDF2 is set to 0 when it is determined as a non-speech section.

  The volume correction means 40 amplifies the received signal with the volume correction amount (amplification degree) instructed from the volume correction amount adjustment means 60. The volume correction amount adjusting unit 60 converts the ambient noise level (estimated value) Pn ′ (n) input from the near-end side ambient noise level estimating unit 3 to the first to fourth reference values XL1 to XL4 (XL4 <XL1 <XL3). The volume correction amount is determined by comparing with <XL2). For example, the sound volume correction amount adjusting means 60 sets the sound volume correction amount to zero (amplification level = 0 dB) when the ambient noise level Pn ′ (n) is smaller than the first reference value XL1, and the ambient noise level Pn ′. When (n) rises and exceeds the first reference value XL1, the volume correction amount is set to 4 dB (amplification level = 4 dB), and the ambient noise level Pn ′ (n) further rises to the second reference value XL2. Is set to 8 dB (amplification level = 8 dB), and on the contrary, if the ambient noise level Pn ′ (n) decreases and falls below the third reference value XL3, the volume correction amount is set to 4 dB. If the ambient noise level Pn ′ (n) further falls below the fourth reference value XL4, the volume correction amount is set to 0 dB. Further, the volume correction amount adjusting means 60 is only set when the far-end side voice section detection flag SDF2 input from the far-end side voice section detecting means 50 is 1 (voice section), and the volume correction amount set at that time ( 0 dB, 4 dB, or 8 dB) is instructed to the volume correction means 40 to perform volume correction.

  Thus, the near-end side ambient noise level estimation means 30 updates the near-end side ambient noise level estimate value Pn ′ (n) when the near-end side speech section detector 34 has not detected a speech section. In addition, since the near-end side ambient noise level estimated value Pn ′ (n) is not updated when the near-end side speech segment detection unit 34 detects a speech segment, the correction in the volume correction means 40 according to the ambient noise level. The volume is adjusted to an appropriate value by the volume correction amount adjusting means 60, and a loud voice call can be made with an appropriate volume according to the level of the ambient noise, and the far-end side voice section detecting means 50 detects the voice section. The volume correction amount adjusting unit 60 adjusts the correction amount in the volume correction unit 40 according to the ambient noise level estimated by the near-end side ambient noise level estimation unit 30 only when the volume is in the non-speech section. Not It is possible to suppress the occurrence of howling due to more wraparound from the speaker 2 to the microphone 1. In addition, since the transmission signal amplifier (received signal level adjustment means) provided in the voice switch VS is used as the volume correction means 40, the configuration is simplified (reduction in program capacity) compared to the case where the volume correction means 40 is provided separately. Can reduce costs. When the volume is corrected by the volume correction means 40 provided in the voice switch VS, there is a possibility that an error may occur in the estimated value of the second instantaneous power estimation unit 8 or the detection result of the far-end side voice section detection means 50. In the second instantaneous power estimation unit 8 and the far-end side voice section detection means 50, the level of the received signal is attenuated by the volume correction amount instructed from the volume correction amount adjustment means 60.

  By the way, if the volume correction means 40 performs volume correction while the voice switch VS is switched to the receiving state, the volume of the sound heard from the speaker 2 may increase or decrease during the call, resulting in a decrease in call quality. There is. Therefore, if the volume correction amount adjustment means 60 does not adjust the correction amount while the voice switch VS is switched to the reception state, the correction amount is adjusted when the voice switch VS is in the idle state. The volume of the sound heard from 2 can be prevented from being lowered or lowered without raising or lowering during the call. Alternatively, if the volume correction amount adjusting means 60 adjusts the correction amount after a predetermined delay time while the voice switch VS is switched to the reception state, the ambient noise level on the near end side frequently fluctuates. Even under circumstances, it is possible to prevent the sound volume heard from the speaker 2 from being raised or lowered in the middle of a call, thereby preventing a reduction in call quality.

  Further, when the volume correction unit 40 performs volume correction while the voice switch VS is switched to the reception state, the near-end side ambient noise level estimation unit 30 estimates by increasing the sneak component from the speaker 2 to the microphone 1. Since the ambient noise level increases and the amount of correction by the sound volume correction means 40 increases as the ambient noise level increases, the ambient noise level may further increase and howling or blocking (receiving blocking) may occur. . Note that reception blocking means that the level of the received signal is lower than the level of the wraparound signal on the near end side (the signal collected by the microphone 1 out of the sound transmitted from the speaker 2), and the utterance from the far end speaker. Even when the voice switch VS is not switched to the receiving state, the voice signal emitted from the far-end speaker is not transmitted from the speaker 2 at a desired level.

  Therefore, if the long-time average value calculation unit 32 does not update the long-time average value Pn while the voice switch VS is switched to the receiving state, the near-end side ambient noise level estimation means 30 estimates the ambient noise level. The value is maintained at the estimated value in the transmission state (or idle state) without changing, and as a result, howling and blocking can be prevented.

  By the way, the voice switch VS detects the detection result of the far-end side voice section detection means 50 (far-end side voice section detection flag SDF2) and the near-end side voice section detection unit 33 constituting the near-end side ambient noise level estimation means 30. The call state may be switched with reference to the detection result (near-end voice section detection flag SDF1). Specifically, not only the output signals C1 and C2 of the first comparator 11 and the second comparator 12, but also these two output signals C1 and C2, the near end side speech section detection flag SDF1, and the far end side speech. The call state may be determined based on the section detection flag SDF2, and there is an advantage that the determination accuracy of the call state is improved. The voice switch VS is provided with means for detecting each voice section on the near end side and the far end side, and those detection means provided on the voice switch VS are used as the near end side voice of the near end side ambient noise level estimation means 30. It may be used as the section detection unit 33 and the far-end voice section detection means 50, and by sharing means having the same function, the cost can be reduced by simplifying the configuration (decreasing the program capacity).

  Further, in order to prevent the above-described reception blocking, as the ambient noise level estimated by the near-end side ambient noise level estimation unit 30 increases, the variable-frequency multiplier 10a included in the near-end side combined gain multiplication unit 10 of the voice switch VS. The coefficient Gr may be increased. That is, the coefficient Gr of the variable coefficient multiplier 10a is a parameter indicating the ease of switching the voice switch VS to the reception state. The larger the value of the coefficient Gr with respect to the same reception signal level, the more the voice switch VS is in the reception state. Therefore, if the coefficient Gr is increased as the ambient noise level increases to make it easier for the voice switch VS to switch to the reception state, occurrence of reception blocking under a situation where the ambient noise level is relatively high can be prevented. it can.

  Alternatively, when the volume correction amount adjusting unit 60 increases the correction amount to increase the volume, the coefficient ηt of the fixed coefficient multiplier 9b included in the far-end feedback gain multiplication unit 9 of the voice switch VS may be decreased. That is, the coefficient ηt of the fixed coefficient multiplier 9b is a parameter indicating the ease of switching the voice switch VS to the transmission state. The smaller the value of the coefficient ηt with respect to the same transmission signal level, the more the voice switch VS becomes. Since the frequency of switching to the transmission state decreases, if the coefficient ηt is decreased as the volume increases to make it difficult for the voice switch VS to switch to the transmission state, the volume is reduced under a situation where the sneaking from the speaker 2 to the microphone 1 is large. Occurrence of incoming call blocking in the case of an increase in the number can be prevented.

(Embodiment 2)
FIG. 3 shows the external structure of the lobby intercom A of this embodiment. The microphone 1, the speaker 2, and each unit described in the first embodiment are housed in a rectangular box-shaped housing 70, and the housing 70 is attached to a wall surface of a common entrance (lobby) of an apartment house such as an apartment.

  Here, in the center of the front surface of the housing 70, a plurality of operation buttons 71 such as a call button for instructing the start of a call and a ten key button for selecting a dwelling unit (dwelling unit number) of a call partner are arranged. The speaker 2 is arranged vertically above the plurality of operation buttons 71, and the microphone 1 is arranged vertically below. If the microphone 1 and the speaker 2 are arranged in this way, the positional relationship between the speaker's ear and the speaker 2 talking on the lobby interphone A and the positional relationship between the speaker's mouth and the microphone 1 are optimized, respectively. 2 can be prevented from occurring due to the acoustic coupling between the microphone 1 and the microphone 1, and the call voice can be transmitted to the speaker's ear at an optimum volume. In addition, since the plurality of operation buttons 71 are disposed in the center of the front surface of the housing 70 between the speaker 2 and the microphone 1, no dead space is generated on the front surface of the housing 70.

  By the way, a general speaker has a structure in which a sound is generated by vibrating a cone-shaped diaphragm, and a sound (sound wave) that is swelled spreads toward the front of the speaker 2, and a microphone is used. In general, since a non-directional microphone is used, the sound generated by the speaker 2 is easily collected by the microphone 1, and the degree of acoustic coupling between the microphone 1 and the speaker 2 increases and howling occurs. There is a fear.

  Therefore, if a plane wave speaker having a structure that vibrates a flat plate-like vibrating body is used as the speaker 2, it is difficult for the sound generated by the speaker 2 to be collected by the microphone 1, and the acoustic coupling between the speaker 2 and the microphone 1 is caused. It is possible to suppress the occurrence of howling and improve call quality. Furthermore, if a microphone having directivity is used as the microphone 1, the sound generated by the speaker 2 becomes difficult to be collected by the microphone 1, and howling caused by acoustic coupling between the speaker 2 and the microphone 1 can be further suppressed. it can. If the microphone 1 has directivity, noise equivalent to the ambient noise that reaches the speaker's ear cannot be collected, and the near-end side ambient noise level estimation accuracy decreases, and the optimum volume level is reduced. 4, microphones 1a, 1b, 1c having a plurality of directivities (three in the illustrated example) are arranged in parallel in the horizontal direction as shown in FIG. If the output of the microphones 1a, 1b, 1c (transmission signal) is added by the adder 80, noise equivalent to the ambient noise reaching the speaker's ear is collected and the volume is corrected to an optimum value. be able to.

It is a block diagram which shows Embodiment 1 of this invention. It is a block diagram which shows the near end side ambient noise level estimation means in the same as the above. It is a front view which shows Embodiment 2 of this invention. It is the schematic which shows the arrangement configuration of the several microphone in the same as the above.

Explanation of symbols

DESCRIPTION OF SYMBOLS 1 Microphone 2 Speaker 30 Near end side ambient noise level estimation means 40 Volume correction means 50 Far end side audio section detection means 60 Volume correction amount adjustment means VS Voice switch

Claims (11)

A microphone and a speaker, and a voice switch for selectively switching between a transmission state in which a transmission signal output from the microphone is transmitted to the far end side and a reception state in which a reception signal transmitted from the far end side is input to the speaker; , A near-end side ambient noise level estimation unit that estimates a near-end side ambient noise level included in the transmission signal, and a volume correction unit that corrects the volume of the sound that the speaker rings by increasing or decreasing the reception signal level; , A far-end side speech section detecting means for detecting a speech section in which the received signal includes a speech component, and a near-end side ambient noise level estimating means when the far-end side speech section detecting means is detecting a speech section And a volume correction amount adjusting means for adjusting a correction amount in the volume correction means according to the ambient noise level estimated in step (b), and the voice switch adjusts the received signal level to increase or decrease the received signal level. And the received signal level adjusting means serves as the volume correction means, and the near-end side ambient noise level estimating means calculates a short-time average value calculation unit for calculating a short-time average value of instantaneous power of the transmission signal. And a long-time average value calculation unit that calculates a long-time average value of the instantaneous power, and the speech signal includes a voice component by comparing the short-time average value and the long-time average value. A near-end side speech section detection unit that detects a section; and an ambient noise level calculation unit that calculates an estimated value of a near-end side ambient noise level included in the transmission signal; Updates the near-end ambient noise level estimate when no speech segment is detected, and updates the near-end ambient noise level estimate when the near-end speech segment detector detects a speech segment a hands-free communication devices that do not, The amount correction amount adjusting means, hands-free communication device while the voice switch is switched to the receiving state, characterized in that not adjusted correction amount. A microphone and a speaker, and a voice switch for selectively switching between a transmission state in which a transmission signal output from the microphone is transmitted to the far end side and a reception state in which a reception signal transmitted from the far end side is input to the speaker; , A near-end side ambient noise level estimation unit that estimates a near-end side ambient noise level included in the transmission signal, and a volume correction unit that corrects the volume of the sound that the speaker rings by increasing or decreasing the reception signal level; , A far-end side speech section detecting means for detecting a speech section in which the received signal includes a speech component, and a near-end side ambient noise level estimating means when the far-end side speech section detecting means is detecting a speech section And a volume correction amount adjusting means for adjusting a correction amount in the volume correction means according to the ambient noise level estimated in step (b), and the voice switch adjusts the received signal level to increase or decrease the received signal level. And the received signal level adjusting means serves as the volume correcting means, and the near-end side ambient noise level estimating means calculates a short-time average value calculating unit for calculating a short-time average value of instantaneous power of the transmitted signal. And a long-time average value calculation unit that calculates a long-time average value of the instantaneous power, and the speech signal includes a voice component by comparing the short-time average value and the long-time average value. A near-end side speech section detection unit that detects a section; and an ambient noise level calculation unit that calculates an estimated value of a near-end side ambient noise level included in the transmission signal, the near-end side speech section detection unit Updates the estimated value of the near-end side ambient noise level when no speech section is detected, and updates the estimated value of the near-end side ambient noise level when the near-end side speech section detector detects a speech section a hands-free communication devices that do not, The amount correction amount adjusting means, expansion voice communication device while the voice switch is switched to the listen state you characterized by adjusting the correction amount after a predetermined delay time. 3. The loudspeaker device according to claim 1, wherein the long-term average value calculation unit does not update the long-term average value while the voice switch is switched to the reception state . The voice communication device according to claim 1 or 2 , wherein the voice switch switches a call state with reference to a detection result of the far-end side voice section detecting means and a detection result of the near-end side voice section detecting unit. . The voice switch includes transmission side loss insertion means for inserting loss into the signal path of the transmission signal, reception side loss insertion means for inserting loss into the signal path of the reception signal, and loss insertion on the transmission side and reception side. Insertion loss amount control means for controlling the loss amount inserted from the means, the insertion loss amount control means, a first instantaneous power estimation unit for estimating the instantaneous power of the input signal to the transmission side loss insertion means, A second instantaneous power estimator for estimating the instantaneous power of the input signal to the receiving-side loss insertion means, and from the input point to the transmitting-side loss insertion means through the transmitting-side loss insertion means and the wraparound on the far-end side A far-end-side feedback gain multiplying means having a value determined according to the gain of the system that feeds back to the input point to the receiving-side loss inserting means, and a receiving-side loss inserting means from the input point to the receiving-side loss inserting means Loss on the sending side after wraparound at the near end Near-end side coupling gain multiplication means having a coefficient determined by the gain of the path to the input point to the insertion means, and near-end side coupling gain multiplication means for the output signal of the second instantaneous power estimation unit A first comparator for comparing the magnitude relationship between the output signal obtained by inputting to the output signal of the first instantaneous power estimation unit and the output signal of the first instantaneous power estimation unit by the far-end feedback gain multiplication A second comparator for comparing the magnitude relationship between the output signal obtained by inputting to the means and the output signal of the second instantaneous power estimator, and the output signals of the first comparator and the second comparator. And an insertion loss amount distribution processing unit that determines the call state and controls the gain of the transmission side loss insertion unit and the reception side loss insertion unit, and the far-end side feedback gain multiplication unit includes the transmission side loss insertion unit. Variable coefficient multiplier with a coefficient approximately equal to the gain of A fixed coefficient multiplier whose coefficient is a value obtained by multiplying the gain of the path from the output point of the stage to the input point of the receiving-side loss insertion means via the far-end side by a predetermined margin value. The end side coupling gain multiplication means includes a variable coefficient multiplier having a coefficient substantially equal to the gain of the reception side loss insertion means, and an input point of the transmission side loss insertion means from the output point of the reception side loss insertion means through the speaker and the microphone. And a fixed coefficient multiplier whose coefficient is a value obtained by multiplying the gain of the route to the predetermined margin by a variable factor multiplier as the ambient noise level estimated by the near-end side ambient noise level estimation means increases 3. The loudspeaker apparatus according to claim 1 or 2 , wherein the coefficient is increased . The voice switch includes transmission side loss insertion means for inserting loss into the signal path of the transmission signal, reception side loss insertion means for inserting loss into the signal path of the reception signal, and loss insertion on the transmission side and reception side. Insertion loss amount control means for controlling the loss amount inserted from the means, the insertion loss amount control means, a first instantaneous power estimation unit for estimating the instantaneous power of the input signal to the transmission side loss insertion means, A second instantaneous power estimator for estimating the instantaneous power of the input signal to the receiving-side loss insertion means, and from the input point to the transmitting-side loss insertion means through the transmitting-side loss insertion means and the wraparound on the far-end side A far-end-side feedback gain multiplying means having a value determined according to the gain of the system that feeds back to the input point to the receiving-side loss inserting means, and a receiving-side loss inserting means from the input point to the receiving-side loss inserting means Loss on the sending side after wraparound at the near end Near-end side coupling gain multiplication means having a coefficient determined by the gain of the path to the input point to the insertion means, and near-end side coupling gain multiplication means for the output signal of the second instantaneous power estimation unit A first comparator for comparing the magnitude relationship between the output signal obtained by inputting to the output signal of the first instantaneous power estimation unit and the output signal of the first instantaneous power estimation unit by the far-end feedback gain multiplication A second comparator for comparing the magnitude relationship between the output signal obtained by inputting to the means and the output signal of the second instantaneous power estimator, and the output signals of the first comparator and the second comparator. And determining the call state and controlling the gain of the transmission-side loss insertion means and the reception-side loss insertion means, and the near-end side combined gain multiplication means includes a reception-side loss insertion means Variable coefficient multiplier with a coefficient approximately equal to gain and insertion loss on the receiver side A fixed coefficient multiplier whose coefficient is a value obtained by multiplying the gain of the path from the output point of the stage through the speaker and microphone to the input point of the transmission side loss insertion means by a predetermined margin value, on the far end side The feedback gain multiplication means includes a variable coefficient multiplier having a coefficient substantially equal to the gain of the transmission side loss insertion means, and the reception side loss insertion means of the reception side loss insertion means through wraparound on the far end side from the output point of the transmission side loss insertion means. A fixed coefficient multiplier whose coefficient is a value obtained by multiplying the gain of the path to the input point by a predetermined margin value. When the volume correction amount adjusting means increases the correction amount to increase the volume, the fixed coefficient multiplication is performed. 3. The loudspeaker apparatus according to claim 1 , wherein a coefficient of the receiver is reduced . 3. The loudspeaker apparatus according to claim 1 , wherein the speaker is a plane wave speaker having a structure for vibrating a flat plate-like vibrating body . The loudspeaker apparatus according to claim 1 or 2 , wherein the microphone is a microphone having directivity . The loudspeaker apparatus according to claim 1 or 2 , further comprising a housing in which a microphone and a speaker are disposed on the front side, and the speaker is disposed vertically above the microphone on the front surface of the housing . 10. The loudspeaker apparatus according to claim 9 , wherein a plurality of types of operation buttons including a call button for instructing the start of a call are disposed between a speaker and a microphone on the front surface of the housing . The loudspeaker apparatus according to claim 9 , wherein the microphone is a plurality of directional microphones arranged in parallel in the horizontal direction .

Images