JP2007036739A - Loudspeaking telephone call apparatus - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To perform a loudspeaking telephone call in suitable volume accopding to the level of peripheral noise and to suppress the occurrence of howling. <P>SOLUTION: A near-end side peripheral noise level estimation means 3 updates an estimation value Pn' of a near-end side peripheral noise level when a sound section is not detected by a near-end side sound section detection part 32, and does not update the estimation value Pn' of the near-end side peripheral noise level when the sound section is not detected. Therefore, correction quantity in a volume correction means 4 is adjusted to a suitable value by a volume correction quantity adjusting means 6 in accordance with the peripheral noise level and loudspeaking can be performed by suitable volume corresponding to the level of the peripheral noise. Since the volume correction quantity adjusting means 6 adjusts the correction quantity of the volume correction means 4 only when a far-end side sound section detection means 5 detects a sound section, volume correction is not performed in a non-sound section and the generation of howling due to runaround from a speaker 2 to a microphone 1 can be suppressed. <P>COPYRIGHT: (C)2007,JPO&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 door phone slave unit, the wraparound component to the microphone is also increased by increasing the speaker volume. Therefore, there is a problem that howling is likely to occur in a situation where the ambient noise is large.

  The present invention has been made in view of the above circumstances, and an object of the present invention is to provide a loudspeaker device that can perform a loudspeaker call at an appropriate volume according to the level of ambient noise and at the same time suppress howling. It is to provide.

  In order to achieve the above object, a first aspect of the present invention provides a microphone and a speaker, and a near-end side ambient noise level estimating means for estimating a near-end side ambient noise level included in a transmission signal output from the microphone. A volume correction means for correcting the volume of the sound that the speaker rings by increasing / decreasing the level of the reception signal input to the speaker, and a voice section in which the received signal before being corrected by the volume correction means includes a voice component. The far-end side speech section detecting means to detect and the correction in the volume correcting means according to the ambient noise level estimated by the near-end side ambient noise level estimating means when the far-end side speech section detecting means is detecting the speech section A volume correction amount adjusting means for adjusting the amount, and the near-end side ambient noise level estimating means, a short-time average value calculating unit for calculating a short-time average value of the instantaneous power of the transmission signal, And a voice interval in which the transmission signal includes a voice component by comparing the short-time average value with the long-time average value. A near-end side speech section detection unit, and an ambient noise level calculation unit that calculates an estimated value of the near-end side ambient noise level included in the transmission signal. Updates the near-end ambient noise level estimate when no section is detected, and does not update the near-end ambient noise level estimate when the near-end speech section detector detects a speech section It is characterized by that.

  According to a second aspect of the present invention, in the first aspect, the volume correction amount adjusting means adjusts the correction amount only when the estimated value of the near-end side ambient noise level estimating means is equal to or greater than a predetermined threshold value. It is characterized by.

  According to a third aspect of the present invention, in the first aspect of the invention, the long-time average value calculation unit increases the average time after the lapse of a predetermined time from the start of calculation with respect to the average time immediately after the calculation of the long-time average value is started It is characterized by doing.

  According to a fourth aspect of the present invention, in the first aspect of the invention, the volume correction amount adjusting means does not adjust the correction amount when the received signal level exceeds a predetermined upper limit value.

  According to a fifth aspect of the present invention, in the first aspect of the invention, the far-end voice section detecting means estimates a received signal level when a voice section is detected, and the volume correction amount adjusting means is the received signal level. The correction amount is adjusted only when the estimated value is less than or equal to a predetermined threshold value.

  According to a sixth aspect of the present invention, in the first aspect of the present invention, the volume correction means relatively reduces the delay time when increasing the received signal level and relatively decreases the delay time when decreasing the received signal level. It is characterized by being enlarged.

  A seventh aspect of the invention is characterized in that, in the first aspect of the invention, a low-pass filter having a cutoff frequency higher than the frequency of the sound component is provided between the volume correction means and the speaker.

  The invention of claim 8 is the invention of claim 1, further comprising a low-pass filter for removing a frequency component higher than a speech component from a transmission signal inputted to the near-end side ambient noise level estimation means.

  According to a ninth aspect of the present invention, in the first aspect of the present invention, a function for transmitting a call signal for calling the other party's communication device instead of the transmission signal and a back tone signal for notifying that the call is in progress are received. And a back tone signal removal filter that removes the back tone signal and is provided between the microphone and the near-end side ambient noise level estimation means.

  According to a tenth aspect of the present invention, in the first aspect of the present invention, the ambient noise that suppresses the ambient noise included in the transmitted signal is further downstream than the near-end-side periodic noise level estimating means in the speech path through which the transmitted signal is transmitted. A suppression means is provided.

  The invention of claim 11 is characterized in that, in the invention of claim 10, the ambient noise suppression means suppresses the ambient noise using the estimated value of the near-end-side periodic noise level estimation means.

  The invention of claim 12 is characterized in that, in the invention of claim 1, the sound volume correction means gradually increases or decreases with a time constant when increasing or decreasing the received signal level.

  According to a thirteenth aspect of the present invention, in the first aspect of the invention, the near-end-side ambient noise level estimation means does not detect that the near-end-side voice section detector detects a voice section in a certain period immediately after the start of a call. Regardless, the estimated value of the near-end side ambient noise level calculated by the ambient noise level calculation unit is updated.

  A fourteenth aspect of the present invention is characterized in that, in the first aspect of the present invention, the speaker is a plane wave speaker having a structure for vibrating a flat plate-like vibrating body.

  The invention of claim 15 is characterized in that, in the invention of claim 1, the microphone is a microphone having directivity.

  A sixteenth aspect of the present invention is characterized in that, in the first aspect of the invention, a housing in which a microphone and a speaker are arranged on the front side is provided, and a speaker is arranged vertically above the microphone on the front side of the housing.

  The invention of claim 17 is the invention of claim 16, 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 18 is characterized in that, in the invention of claim 16, the microphones are 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 cause the howling.

  According to the second aspect of the present invention, the volume of the speaker does not increase excessively under a situation where the ambient noise level is low, and an unpleasant sound is prevented from ringing from the speaker, thereby improving the call quality.

  According to the third aspect of the present invention, the average value for a long time is calculated with a relatively short average time immediately after the start of a call, and the average value for a long time is calculated with a relatively long average time after a predetermined time has elapsed. The sound volume can be optimally corrected while shortening the time required to calculate the estimated value of the ambient noise level.

  According to the invention of claim 4, the volume of the sound is not increased excessively, and the sound quality is prevented by preventing sound cracking and the like.

  According to the fifth aspect of the present invention, it is possible to prevent a decrease in call quality due to an excessive increase in volume.

  According to the sixth aspect of the present invention, even when the ambient noise level changes suddenly, the volume can be corrected without causing discomfort to the speaker, and a comfortable call can be made.

  According to the invention of claim 7, by suppressing the frequency component higher than the voice component, it is possible to suppress the occurrence of howling due to the acoustic coupling between the speaker and the microphone, and the call quality is improved.

  According to the eighth aspect of the present invention, it is possible to perform volume correction in accordance with ambient noise in a frequency band close to the frequency band of voice, thereby improving call quality.

  According to the ninth aspect of the present invention, it is possible to estimate the ambient noise level without being affected by the back tone signal, so that it is possible to perform volume correction immediately after the actual call starts.

  According to the invention of claim 10, since the ambient noise contained in the transmitted signal is suppressed by the ambient noise suppressing means, both the received voice and the transmitted voice become clear.

  According to the eleventh aspect, the configuration can be simplified.

  According to the twelfth aspect of the present invention, it is possible to eliminate discomfort associated with an increase or decrease in volume during a call.

  According to the invention of claim 13, to update the estimated value of the near-end side ambient noise level calculated by the ambient noise level calculation unit regardless of whether or not a voice section is detected for a certain period immediately after the start of the call. The calculation of the estimated value of the ambient noise level will not be delayed.

  According to the fourteenth 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 fifteenth aspect of the present invention, it is difficult for the sound generated 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 invention of claim 16, 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 seventeenth aspect of the present invention, no dead space is generated on the front surface of the housing.

  According to the eighteenth aspect of the present invention, it is possible to collect noise equivalent to the ambient noise reaching the speaker's ear and correct the 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, the near-end side ambient noise level estimating means 3 for estimating the near-end side ambient noise level included in the transmission signal output from the microphone 1, and the speaker 2 are input. Volume correction means 4 for correcting the volume of the sound that the speaker 2 rings by increasing / decreasing the reception signal level, and a far distance for detecting a voice section in which the reception signal before being corrected by the volume correction means 4 includes a voice component In the volume correction unit 4 according to the ambient noise level estimated by the near-end side ambient noise level estimation unit 3 when the end-side speech segment detection unit 5 and the far-end side speech segment detection unit 5 detect the speech segment. Volume correction amount adjusting means 6 for adjusting the correction amount is provided. In the present embodiment, each of the near-end side ambient noise level estimation means 3, the volume correction means 4, the far-end side speech section detection means 5, and the volume correction amount adjustment means 6 is replaced with a DSP (digital signal processor), This is realized by controlling hardware such as a 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).

  As shown in FIG. 2, the near-end side ambient noise level estimation means 3 includes a short-time average value calculation unit 30 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 31 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 32 and an ambient noise level calculation unit 33 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 30 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 31 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 32 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 calculator 33 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 transmitted 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 3, the far-end side speech section detection unit 5 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 unit 4 amplifies the received signal with the volume correction amount (amplification degree) instructed from the volume correction amount adjustment unit 6 and outputs the amplified signal to the speaker 2. The volume correction amount adjusting unit 6 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 unit 6 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 6 is only set when the far end side speech section detection flag SDF2 input from the far end side speech section detecting means 5 is 1 (speech section) (the volume correction amount set at that time ( 0 dB, 4 dB, or 8 dB) is instructed to the volume correction means 4 to perform volume correction.

  Thus, the near-end side ambient noise level estimation means 3 updates the near-end side ambient noise level estimate value Pn ′ (n) when the near-end side speech segment detector 33 does not detect a speech segment. At the same time, when the near-end side speech section detection unit 33 detects a speech section, the estimated value Pn ′ (n) of the near-end side ambient noise level is not updated, so that the volume correction means 4 performs correction according to the ambient noise level. The volume is adjusted to an appropriate value by the volume correction amount adjusting means 6, and a loud voice call can be made with an appropriate volume according to the level of ambient noise. Further, the far-end voice section detecting means 5 detects the voice section. Only when the volume correction amount adjustment means 6 adjusts the correction amount in the volume correction means 4 in accordance with the ambient noise level estimated by the near-end side ambient noise level estimation means 3, the volume correction is performed in the non-voice section. Not by speaker It is possible to suppress the occurrence of howling due to wraparound to the microphone 1 from. If the volume correction amount adjustment means 6 adjusts the correction amount only when the estimated value Pn ′ (n) of the near-end side ambient noise level estimation means 3 is greater than or equal to a predetermined threshold value, the ambient noise can be adjusted. In a situation where the level is low, the volume of the speaker 2 does not become excessively high, and an unpleasant sound can be prevented from being emitted from the speaker 2 to improve the call quality.

Here, in the long-term average value calculation unit 31 of the near-end side ambient noise level estimation means 3, the average time T1 (= ρ2 ₁ / k, where k is the sampling rate) immediately after the calculation of the long-term average value Pn is started. The average time T2 (= ρ2 ₂ / k) after the elapse of a predetermined time from the start of calculation is lengthened, and the long-term average value Pn is calculated at a relatively short average time T1 immediately after the start of the call. If the long-term average value Pn is calculated with a relatively long average time T2 after the lapse of time, it is possible to perform optimal sound volume correction while shortening the time required to calculate the estimated value Pn ′ of the ambient noise level.

  By the way, when the received signal level is high, if the volume correction unit 4 corrects the volume, the volume may become too high, resulting in sound cracking and the like, which may reduce the call quality. Therefore, the reception signal level (short-term average value of the reception signal) when the voice section is detected by the far-end side voice section detection means 5 is estimated, and the volume correction amount adjustment means 6 sets the reception signal level to a predetermined upper limit. If the value exceeds the value, the correction amount is not adjusted. That is, the volume correction amount adjusting unit 6 is only adjusted when the amplification degree of the volume correction unit 4 is saturated or the estimated value of the received signal level is equal to or less than a predetermined threshold value. However, if the correction amount is adjusted, the volume will not be increased too much, and sound quality will be prevented by improving sound quality.

  Also, in the volume correction means 4, the volume correction amount is actually increased or decreased (for example, from 0 dB to 4 dB, from 4 dB to 8 dB, or from 8 dB to 4 dB) when the ambient noise level exceeds or falls below the reference value XL1,. A time difference (delay time) from the point of change) and a relatively small delay time when increasing the received signal level and a relatively large delay time when decreasing the received signal level. For example, even when the ambient noise level changes abruptly, the volume can be corrected without causing discomfort to the speaker, and a comfortable call can be made.

  Furthermore, if the volume correction unit 4 increases or decreases the volume correction amount (for example, changes from 0 dB to 4 dB, 4 dB to 8 dB, or 8 dB to 4 dB) and gradually increases or decreases the time constant, The discomfort associated with the increase or decrease of can be eliminated.

  By the way, the average value Pn for a long time is not fixed immediately after the start of the call, and therefore the voice section cannot be detected with high accuracy by the near-end voice section detection unit 32. Therefore, the estimated value Pn ′ of the ambient noise level There is a possibility that the calculation of (n) is delayed. Therefore, in a certain period immediately after the start of the call, the near-end side ambient noise level calculated by the ambient noise level calculation unit 33 is not related to whether or not the near-end side voice segment detection unit 32 detects the voice segment. If the estimated value Pn ′ (n) is updated, the calculation of the estimated value of the ambient noise level can be prevented from being delayed.

(Embodiment 2)
As shown in FIG. 3, the present embodiment has a cutoff frequency higher than the frequency of the audio component, the low-pass filter 7 provided between the volume correction means 4 and the speaker 2, and the frequency of the audio component. A low-pass filter 8 that has a high cutoff frequency and removes a frequency component higher than a speech component from a transmission signal input to the near-end side ambient noise level estimation means 8 is characterized.

  That is, by suppressing the frequency component higher than the voice component by the low-pass filter 7, there is an advantage that howling due to acoustic coupling between the speaker 2 and the microphone 1 can be suppressed and the call quality is improved. Further, since the transmission signal is input to the near-end side ambient noise level estimation means 3 through the low-pass filter 8, the volume can be corrected according to the ambient noise in the frequency band close to the frequency band of the voice, and the call can be performed. There is an advantage that quality is improved.

(Embodiment 3)
In general, a lobby interphone has a function of sending a call signal for calling a communication device on the other side (such as a dwelling unit installed in each dwelling unit of an apartment house) and a back tone signal for notifying that a call is being made. 2 is provided. However, there is a possibility that a back tone (for example, a chime sound) ringed from the speaker 2 will enter the microphone 1 and affect the estimated value of the ambient noise level.

  Therefore, in the present embodiment, as shown in FIG. 4, a backtone signal removal filter 9 for removing the backtone signal is provided between the microphone 1 and the near-end side ambient noise level estimation means 3. For example, if the back tone signal has frequency characteristics having peaks at frequencies f1, f2, and f3 as shown in FIG. 5A, the back tone signal has frequencies f1 to f3 as shown in FIG. The backtone signal removal filter 9 may be configured by a notch filter having a frequency characteristic that attenuates the range.

  Therefore, since the frequency component of the back tone signal is removed from the transmission signal input to the near-end side ambient noise level estimation means 3 by the back tone signal removal filter 9, the back tone is produced from the speaker 2. The ambient noise level can be estimated without being affected by the back tone signal, and the volume correction can be performed immediately after the actual call starts.

(Embodiment 4)
In the present embodiment, as shown in FIG. 6 (a), in the speech path where the transmission signal is transmitted, the ambient noise included in the transmission signal is suppressed downstream of the near-end side periodic noise level estimation means 3. It is characterized in that the noise suppression means 10 is provided.

  As shown in FIG. 6B, the ambient noise suppression unit 10 includes a Fourier transform unit 10a that performs Fourier transform (discrete Fourier transform) on an input signal (transmission signal), and a real part (amplitude component) of the input signal after Fourier transform. ) To determine a level of a frequency component (noise component) regarded as ambient noise, and a gain function calculation unit 10c to obtain a gain function for attenuating the noise component obtained by the noise component calculation unit 10b. The noise suppression calculation unit 10d attenuates the noise component by the gain function obtained by the gain function calculation unit 10c, and the inverse of the input signal in which the noise component is attenuated by the noise suppression calculation unit 10d is inverse Fourier transformed (discrete inverse Fourier transformation). And a Fourier transform unit 10e.

  That is, the ambient noise suppression means 10 converts the input signal (transmission signal) from the time domain to the frequency domain by the Fourier transform unit 10a, and then sets the level of the frequency component (noise component) regarded as ambient noise to the noise component calculation unit 10b. The gain function is calculated from the real part (amplitude component) of the input signal after the Fourier transform, and the gain function calculating unit 10c determines a gain function for attenuating the noise component according to the level for each frequency regarded as ambient noise, and the noise suppression calculating unit 10d. A gain component is used to obtain an amplitude component that suppresses the noise component from the real part of the input signal, and the inverse Fourier transform unit 10e performs an inverse Fourier transform on the input signal so that the ambient noise is suppressed ( (Transmission signal) is output.

  As described above, according to the present embodiment, the ambient noise contained in the transmitted signal is suppressed by the ambient noise suppression means 10, so that both the received voice and the transmitted voice become clear. Note that if the ambient noise suppression unit 10 is configured to suppress ambient noise using the estimated value Pn ′ of the near-end-side periodic noise level estimation unit 3, the overall configuration (or a program for controlling DSP or CPU hardware) Can be simplified.

(Embodiment 5)
FIG. 7 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 to fourth embodiments are housed in a rectangular box-shaped housing 20, and the housing 20 is attached to a wall surface of a common entrance (lobby) of an apartment house such as an apartment. .

  Here, a plurality of operation buttons 21 such as a call button for instructing the start of a call and a numeric keypad for selecting a caller's dwelling unit (dwelling unit number) are arranged at the front center of the housing 20. The speaker 2 is disposed vertically above the plurality of operation buttons 21 and the microphone 1 is disposed 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 21 are arranged in the center of the front surface of the housing 20 between the speaker 2 and the microphone 1, no dead space is generated on the front surface of the housing 20.

  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. Therefore, as shown in FIG. 8, a plurality of (three in the illustrated example) directional microphones 1a, 1b, 1c are arranged in parallel in the horizontal direction. If the outputs (transmission signals) of the microphones 1a, 1b, and 1c are added by the adder 50, noise equivalent to 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 block diagram which shows Embodiment 2 of this invention. It is a block diagram which shows Embodiment 3 of this invention. (A) is a figure which shows the frequency characteristic of a back tone signal, (b) is a figure which shows the frequency characteristic of a back tone signal removal filter, respectively. (A) is a block diagram which shows Embodiment 4 of this invention, (b) is a block diagram of an ambient noise suppression means. It is a front view which shows Embodiment 5 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 3 Near end side ambient noise level estimation means 4 Volume correction means 5 Far end side audio section detection means 6 Volume correction amount adjustment means

Claims (18)

A microphone and a speaker, a near-end side ambient noise level estimating means for estimating a near-end side ambient noise level included in a transmission signal output from the microphone, and a speaker by increasing or decreasing a received signal level input to the speaker Volume correction means for correcting the volume of the sound to be sounded, far-end side voice section detection means for detecting a voice section in which the received signal before the correction by the volume correction means includes a voice component, and far-end side voice A volume correction amount adjusting unit that adjusts a correction amount in the volume correction unit according to the ambient noise level estimated by the near-end side ambient noise level estimation unit when the section detection unit is detecting a voice segment;
The near-end side ambient noise level estimation means includes a short-time average value calculating unit that calculates a short-time average value of instantaneous power of the transmission signal, and a long-time average value calculating unit that calculates a long-time average value of the instantaneous power A near-end side speech section detecting unit for detecting a speech section in which the transmission signal includes a speech component by comparing the short-time average value and the long-time average value, and included in the transmission signal An ambient noise level calculation unit that calculates an estimated value of the near-end side ambient noise level, and the near-end side ambient noise level estimate value when the near-end side speech segment detection unit does not detect a speech segment. A loudspeaker apparatus that is updated and does not update the estimated value of the near-end side ambient noise level when the near-end side speech section detection unit detects a speech section.   2. The loudspeaker apparatus according to claim 1, wherein the volume correction amount adjusting means adjusts the correction amount only when the estimated value of the near-end side ambient noise level estimating means is equal to or greater than a predetermined threshold value.   2. The loudspeaker device according to claim 1, wherein the long-time average value calculation unit extends an average time after a predetermined time has elapsed from the start of calculation with respect to an average time immediately after starting calculation of the long-time average value. .   The loudspeaker apparatus according to claim 1, wherein the volume correction amount adjusting means does not adjust the correction amount when the received signal level exceeds a predetermined upper limit value.   The far-end side voice section detecting means estimates the received signal level when the voice section is detected, and the volume correction amount adjusting means is only when the estimated value of the received signal level is equal to or less than a predetermined threshold value. The loudspeaker apparatus according to claim 1, wherein a correction amount is adjusted.   2. The loudspeaker call according to claim 1, wherein the volume correction means relatively decreases a delay time when the received signal level is increased and relatively increases a delay time when the received signal level is decreased. apparatus.   The loudspeaker apparatus according to claim 1, wherein a low-pass filter having a cutoff frequency higher than the frequency of the voice component is provided between the volume correction means and the speaker.   2. The loudspeaker apparatus according to claim 1, further comprising a low-pass filter that removes a frequency component higher than a voice component from a transmission signal input to the near-end side ambient noise level estimation means. A function of sending a call signal for calling the other party's call device instead of a transmission signal and a function of sending a back tone signal for notifying that a call is being made to a speaker instead of a reception signal;
The loudspeaker apparatus according to claim 1, wherein a backtone signal removal filter for removing the backtone signal is provided between the microphone and the near-end side ambient noise level estimation means.   2. An ambient noise suppression means for suppressing ambient noise included in the transmission signal is provided downstream of the near-end-side periodic noise level estimation means in the speech path through which the transmission signal is transmitted. The loudspeaker device described.   11. The loudspeaker apparatus according to claim 10, wherein the ambient noise suppression unit suppresses the ambient noise using the estimated value of the near-end-side periodic noise level estimation unit.   2. The loudspeaker apparatus according to claim 1, wherein the sound volume correcting means gradually increases or decreases with a time constant when increasing or decreasing the received signal level.   The near-end side ambient noise level estimation means calculates the near-end noise level calculation unit in the fixed period immediately after the start of the call, regardless of whether the near-end side voice segment detection unit detects a voice segment. The loudspeaker apparatus according to claim 1, wherein the estimated value of the side ambient noise level is updated.   2. 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, wherein the microphone is a microphone having directivity.   The loudspeaker apparatus according to claim 1, 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.   The loudspeaker apparatus according to claim 16, 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.   The loudspeaker apparatus according to claim 16, wherein the microphone is a plurality of directional microphones arranged in parallel in the horizontal direction.

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