JP4179224B2 - Intercom equipment - Google Patents

Description

  The present invention relates to an intercom device having a recording function for recording a voice of a call with a call device of the other party, a message for listening to the call device of the other party, and the like.

Conventionally, intercom equipment (main unit) installed in each dwelling unit of an apartment like a condominium, intercom slave units installed outside each dwelling unit (for example, at the entrance), and lobby intercom units installed at common entrances In the intercom system that consists of the above, there is a function to send a response message recorded in advance when a call is made from the intercom handset or lobby intercom through the speaker of the calling device, In some cases, the master unit has a function of recording a visitor's message or voice during a call (see Patent Document 1).
JP 2002-57803 A

  However, when recording a response message or transmission voice in the main unit, recording a reception voice from the intercom handset, and recording a reception voice from the lobby interphone, the signal level of the voice signal is different. As a result, there is a difference in the volume of the recorded sound. When the recorded sound is played back, it is difficult to hear the sound when the recorded sound is played back.

  The present invention has been made in view of the above circumstances, and an object of the present invention is to provide an interphone device that can adjust the sound volume during playback to an appropriate level by adjusting the sound volume during recording.

In order to achieve the above object, the invention of claim 1 is an interphone device that is installed in a house and performs a two-way loudspeaking call through a communication path that is appropriately formed with a telephone device installed outside the house. In addition, a voice switch for passing only one of a microphone and a speaker, a transmission signal collected by the microphone, and a reception signal sent from the other party's telephone device , and a voice switch are provided between the microphone and the speaker. The first echo canceller, the second echo canceller provided between the voice switch and the telephone device, and a voice call processing means for realizing a two-way voice call, and a transmission output by the first echo canceller signal and a second recording means for recording a signal mixing the transmission signal and the reception signal together with the echo canceller to record received signal to be output A first volume control means for adjusting the sound volume of the reception signals sent to the recording unit, and a second volume control means for adjusting the volume of the transmission signals sent to the recording unit, transmission signal and the reception signal is mixed Third volume adjusting means for adjusting the volume of the received signal, fourth volume adjusting means for adjusting the volume of the transmitted signal before mixing, and fifth volume adjustment for adjusting the volume of the received signal before mixing Means, volume adjustment value storage means for storing the volume adjustment value of at least one of the transmission signal and the reception signal for each of a plurality of types of telephone equipment, and depending on the type of the other telephone equipment when recording by the recording means Volume adjustment value setting means for reading out the volume adjustment value from the volume adjustment value storage means and setting it in the first to fifth volume adjustment means is provided.

According to the present invention, by adjusting the volume of the received signal by the first volume adjusting means, it is possible to record at an appropriate volume without depending on the volume of the received signal by the other party's call device , , by adjusting the volume of the transmission signal at a second volume control means, since Ru can record a proper volume in accordance with the contents to be recorded can be playback volume to a proper level. Furthermore, since the transmission signal and the reception signal in which the acoustic echo and the line echo are suppressed by the first and second echo cancellers are recorded, the quality of the recorded voice is improved, and the transmission signal and the reception signal before mixing are further improved. The volume of the sound is adjusted by the fourth and fifth volume adjusting means, and the volume of the mixed signal is adjusted by the third volume adjusting means. Can be easily adjusted.

According to a second aspect of the present invention, in the first aspect of the invention, the silent period detecting means for detecting a silent period that can be regarded as having no voice of the signal recorded by the recording means, and the silent period detected by the silent period detecting means And a silent period attenuation means for attenuating a signal recorded by the recording means during the period.

  According to the present invention, by attenuating a signal during a non-voice period that can be regarded as having no voice, unnecessary sounds other than voice such as background noise are not recorded, so that the voice can be clearly reproduced during reproduction.

According to a third aspect of the present invention, in the first aspect of the invention, the silent period detecting means for detecting a silent period that can be regarded as having no sound of the signal recorded by the recording means, and the recording means for the silent period detected by the silent period detecting means And a silent period attenuating means for attenuating the signal to be recorded.

  According to the present invention, since an unnecessary sound is not recorded by attenuating a silent period signal that can be regarded as having no sound, the sound can be clearly reproduced during reproduction.

According to a fourth aspect of the present invention, in the second or third aspect of the invention, the silent period attenuating means or the silent period detecting means adjusts the amount of attenuation according to the type of the partner telephone equipment.

  According to this invention, it is possible to adjust to an appropriate attenuation amount according to the difference in background noise level depending on the other party's telephone equipment.

According to a fifth aspect of the present invention, in the first aspect of the present invention, the silent period detecting means for detecting a silent period in which there is no voice of the signal recorded by the recording means, and the silent period detected by the silent period detecting means And removing means for removing the period from the signal recorded by the recording means.

  According to the present invention, the memory capacity required for recording can be reduced.

According to a sixth aspect of the present invention, in the first aspect of the invention, the silent period detecting means for detecting the silent period that can be regarded as having no sound of the signal recorded by the recording means, and the silent period detected by the silent period detecting means are recorded by the recording means. And removing means for removing from the signal to be recorded.

  According to the present invention, the memory capacity required for recording can be reduced.

According to a seventh aspect of the present invention, in the first aspect of the present invention, a noise removing means for removing a noise component constantly included in a signal recorded by the recording means is provided.

  According to the present invention, since unnecessary noise is not recorded, the sound can be clearly reproduced at the time of reproduction.

According to the present invention, since it is possible to record at an appropriate volume depending on the content to be recorded, without depending on the difference in the volume of the received signal by the other party's call device, the playback volume is set to an appropriate level. It can also, by adjusting the volume of the transmission signal at a second volume control means, since Ru can record a proper volume in accordance with the contents to be recorded, can playback volume to a proper level Further, since the transmission signal and the reception signal in which the acoustic echo and the line echo are suppressed by the first and second echo cancellers are recorded, the quality of the recorded voice is improved, and the transmission signal and the reception signal before mixing are further improved. The volume of the signal can be adjusted to an appropriate level by adjusting the volume of the signal with the fourth and fifth volume adjusting means and adjusting the volume of the mixed signal with the third volume adjusting means. Also, the volume adjustment at the time of recording there is an effect that easily performed.

  Hereinafter, an embodiment in which the present invention is applied to an interphone master unit (hereinafter abbreviated as “master unit”) used in an interphone system as described in the conventional example will be described in detail with reference to the drawings.

Before describing an embodiment of the present invention, a reference example having a basic configuration in common with the embodiment of the present invention will be described.
As shown in FIG. 1, the base unit M, which is an interphone device of this reference example , is connected to a microphone 1 and a speaker 2 and a telephone equipment such as an interphone handset or a lobby interphone (not shown) connected by two telephone lines. A four-line conversion unit 3, a voice call processing unit 4 provided in the middle of a transmission signal path from the microphone 1 to the two-wire four-line conversion unit 3 and a reception signal path from the two-line four-line conversion unit 3 to the speaker 2; A microphone amplifier 5 for amplifying a speech signal collected by the microphone 1, an A / D converter 6 for converting an analog speech signal amplified by the microphone amplifier 5 into a digital speech signal, and a speech communication processing unit D / A converter 7 for converting the digital transmission signal processed in step 4 into an analog transmission signal, and the analog transmission signal output from D / A converter 7 is amplified. The transmission side line amplifier 8, the reception side line amplifier 9 that amplifies the analog reception signal sent through the communication line from the other party's communication device, and the analog reception signal amplified by the reception side line amplifier 9 are digitally converted. An A / D converter 10 that converts the received signal into a received signal, a D / A converter 11 that converts the digital received signal processed by the voice call processing unit 4 into an analog received signal, and an analog output from the D / A converter 11 A speaker amplifier 12 that amplifies the received signal and outputs it to the speaker 2, a recording unit 20 that performs signal processing for recording the transmitted signal and the received signal (hereinafter referred to as “call signal”), and the recording unit 20. And a data storage unit 30 for storing the signal-processed call signal.

  The loudspeaker call processing unit 4 has an unpleasant echo (acoustic echo or acoustic echo) by a return path on the acoustic side formed by acoustic coupling of the microphone 1 and the speaker 2 or a return path on the line side formed with the other party's telephone device. (Line echo) is heard, or a closed loop in which the loop gain in an arbitrary frequency component exceeds 1 times is formed in the communication system due to the feedback path or the like, and howling occurs at the frequency. For this purpose, first and second echo cancellers EC1 and EC2 and a voice switch VSW are provided.

Although not shown, the first echo canceller EC1 has a conventionally known configuration including an adaptive filter and a subtractor, and an impulse of a feedback path (acoustic echo path) formed by acoustic coupling between the speaker 2 and the microphone 1. The response is adaptively identified by the adaptive filter, and the acoustic echo is suppressed by subtracting the pseudo echo component (acoustic echo) estimated from the reference signal (output signal to the speaker 2) from the input signal from the microphone 1 by the subtractor. To do. The second echo canceller EC2 also has a conventionally known configuration including an adaptive filter and a subtracter, and has a reflection due to impedance mismatch between the two-wire / four-wire conversion unit 3 and the communication line, and the other telephone device ( The impulse response of the feedback path (line echo path) formed by the acoustic coupling between the speaker and the microphone in the interphone slave unit or lobby interphone is adaptively identified by the adaptive filter, and the reference signal (2-wire 4-wire conversion unit) The line echo is suppressed by subtracting the pseudo echo component (line echo) estimated from the output signal to 3) from the output signal (received signal) from the 2-wire 4-wire conversion unit 3 by a subtractor. The voice switch VSW also has a conventionally known configuration, and always estimates the call state (sending state, receiving state) from the signal level of the transmission signal and the reception signal, and for example, inserts a loss into one of the signal paths. By doing so, the call state is switched. Note that the first and second echo cancellers EC1 and EC2 and the voice switch VSW in this reference example control the DSP (digital signal processor) hardware by the software (program) for the echo canceller and the voice switch. Is realized.

The recording unit 20 includes a recording / playback signal processing unit 21 that records a call signal, a first volume adjustment unit 22 that adjusts the volume of a digital reception signal sent to the recording / playback signal processing unit 21, and a recording / playback signal processing unit. The second volume adjusting unit 23 that adjusts the volume of the digital transmission signal sent to 21 and the transmission signal output from the first echo canceller EC1 and the reception signal output from the second echo canceller EC2 are mixed. It has a mixing unit 24 and a third volume adjusting unit 25 that adjusts the volume of a signal obtained by mixing the transmission signal and the reception signal (hereinafter referred to as “mixing call signal”). The recording / playback signal processing unit 21 compresses the call signal and the mixing call signal by a compression method such as ADPCM (Adaptive Differential Pulse Code Moudulation) and rewrites the compressed data (call data) from a non-volatile memory (such as EEPROM). The data storage unit 30 is written and stored (recorded). The recording unit 20 in this reference example uses a DSP common to the first and second echo cancellers EC1 and EC2 and the voice switch VSW, and controls the DSP hardware with dedicated software (program). It is realized by. Although detailed description is omitted, the recording / playback signal processing unit 21 reads out and expands the call data stored (recorded) in the data storage unit 30 and transmits the expanded call signal or mixing call signal to the receiving side signal path or It also has a playback function for playing back the recorded voice by sending it to the transmitter signal path.

Here, the base unit M of this reference example outputs a pre-recorded response message from the speaker of the calling device when a visitor cannot answer a call from the calling device by a visitor due to absence or busyness. A function for recording a message left by a visitor in response to the response message (hereinafter referred to as “answering function”) and a function for recording a call (conversation) with a visitor by a calling device (hereinafter referred to as “call recording function”). "). When recording the response message, the volume of the response message (sending signal) is adjusted to an appropriate level by the second volume adjusting unit 23 and then compressed and recorded (recorded) by the recording / playback signal processing unit 21. When recording a visitor's message, the recording / playback signal processing unit 21 adjusts the volume of the message (received signal) to an appropriate level according to the type of the telephone device by the first volume adjustment unit 22. Compressed and recorded (recorded), and when recording the conversation between the visitor and the housekeeper, the volume of the conversation (mixing call signal) is adjusted to an appropriate level according to the calling device by the third volume adjustment unit 25. Then, it is compressed and recorded (recorded) by the recording / playback signal processing unit 21.

Thus, in the base unit M of this reference example , it is possible to record the message at the time of the answering machine without depending on the difference in the volume of the received signal by the other party's telephone device, and the answering machine function. Since the response message can be recorded at an appropriate volume, the volume of the response message or message can be reproduced at an appropriate level. In addition, when recording a call, the mixed call signal in which acoustic echo and line echo are suppressed by the first and second echo cancellers EC1 and EC2 is recorded, so that the quality of the recorded voice is improved and further mixed. By adjusting the volume of the signal by the third volume adjusting unit 25, the volume during reproduction can be set to an appropriate level.

(Embodiment 1 )
As shown in FIG. 2, the basic unit M of the present embodiment has the same basic configuration as that of the reference example . Therefore, the same components are denoted by the same reference numerals and description thereof is omitted.

When recording a call with a visitor using a call device using the call recording function described in the reference example , the call is usually sent from the other call device compared to the volume of the transmitted signal collected by the microphone 1. The volume of the incoming signal is often lower, and there may be an unpleasant sensation of hearing when the volume is increased or decreased during playback, but the transmitted signal and the received signal are mixed. After that, the volume of the transmitted signal and the received signal cannot be made uniform only by adjusting the volume by the third volume adjusting unit 25.

  Therefore, in the present embodiment, the recording unit 20 includes the fourth volume adjustment unit 26 that adjusts the volume of the transmission signal before mixing, and the fifth volume adjustment unit 27 that adjusts the volume of the reception signal before mixing. The volume of the transmission signal and the reception signal before mixing is individually adjusted by the fourth and fifth volume adjustment units 26 and 27. Therefore, even when there is a difference in volume between the transmission signal and the reception signal, the volume is adjusted after adjusting the volume to the same level, and then mixed, and the volume of the mixed call signal (mixed call signal) is set to the third volume adjustment unit 25. Therefore, the sound volume during reproduction can be stabilized at an appropriate level, and there is no discomfort in the sense of hearing due to volume fluctuation.

  Here, regarding the volume adjustment level (volume adjustment value) in the first to fifth volume adjustment units 22, 23, 25 to 27, the type of the other party's call device (for example, interphone slave unit and lobby interphone) Since it is considered that the appropriate level varies depending on the contents to be recorded (response message and message), in this embodiment, the volume adjustment values of the transmission signal and the reception signal are set as parameters for each of a plurality of types of telephone equipment or for each recording content. The parameter storage unit 31 serving as the stored volume adjustment value storage means and the volume adjustment value corresponding to the type of the other party's telephone device and the recording content when the recording unit 20 records are read out from the parameter storage unit 31 to the first to fifth. And a control unit 32 as volume adjustment value setting means for setting the volume adjustment units 22, 23, 25 to 27.

  The parameter storage unit 31 is composed of a rewritable nonvolatile memory (EEPROM, etc.) and the like, and optimal volume adjustment values of the transmission signal and the reception signal are stored as parameters for each of a plurality of types of telephone devices and recording contents. . The control unit 32 includes a CPU, a memory, and the like, and realizes the control of a one-chip DSP that realizes the voice call processing unit 4 and the recording unit 20, the above-mentioned answering function and call recording function, and the like. The volume adjustment value stored in the parameter storage unit 31 is read out when the response message or message is recorded in the answering function and when the call signal is recorded in the call recording function, and is given to the recording unit 20 comprising the DSP. The respective volume adjustment values are set in the five volume adjustment sections 22, 23, 25 to 27.

  As described above, according to the present embodiment, the optimum volume adjustment values for the first to fifth volume adjustment units 22, 23, 25 to 27 are stored in the parameter storage unit 31, and the control unit 32 is used as necessary. However, by setting the volume adjustment values read from the parameter storage unit 31 in the first to fifth volume adjustment units 22, 23, 25 to 27, there is an advantage that the volume adjustment during recording can be easily performed.

(Embodiment 2 )
As shown in FIG. 3, the base unit M of the present embodiment has the same basic configuration as that of the reference example and the first embodiment. Therefore, common constituent elements are denoted by the same reference numerals and description thereof is omitted.

  In the present embodiment, a speechless period detection unit 28 that detects a speechless period in which a speech signal or a mixed speech signal can be regarded as having no voice, and a speech signal or mixing call in a silent period detected by the silent period detection unit 28 There is a feature in that the recording unit 20 includes a silent period attenuation processing unit 29 that attenuates a signal.

  The silent period detection unit 28 estimates the instantaneous power of the call signal (sending signal or received signal) or the mixing call signal and the background noise power included in the call signal or the mixing call signal, respectively, The logical product of the comparison result with the predetermined threshold value and the comparison result obtained by comparing the ratio of the instantaneous power estimation value and the background noise power estimation value (= instantaneous power estimation value / background noise power estimation value) with the predetermined reference value If the instantaneous power estimate is greater than the threshold and the ratio between the instantaneous power estimate and the background noise power estimate is greater than the reference value, the speech period is determined. Judgment. Here, the threshold value is a threshold value that defines the minimum level of the audio signal, and the reference value is a value that defines the minimum ratio between the audio signal level and the background noise level.

  The silent period attenuation processing unit 29 outputs the call signal or the mixed call signal as it is to the recording / playback signal processing unit 21 when the silent period detection unit 28 does not detect the silent period (during the voice period). When the silent period detecting unit 28 detects the silent period (during the silent period), the call signal or the mixing call signal is attenuated and output to the recording / playback signal processing unit 21. The silent period detection unit 28 and the silent period attenuation processing unit 29 are both realized by using a DSP common to the recording unit 20 and controlling the DSP hardware by software (program).

  Thus, in the present embodiment, the silent period detection unit 28 detects the silent period of the call signal or mixing call signal to be recorded, and the silent period attenuation processing unit 29 detects the speech signal or mixing call of the silent period. Since the signal is attenuated, unnecessary sounds such as background noise are not recorded, and the sound can be clearly reproduced during reproduction. Here, if the attenuation amount when the speech signal or mixing call signal is attenuated by the silent period attenuation processing unit 29 is adjusted in accordance with the type of the other party's telephone device or the content of the recording, it depends on the other party's telephone device and the content of the recording. The attenuation can be adjusted to an appropriate amount without depending on the difference in the background noise level. At this time, an appropriate amount of attenuation corresponding to the type of the other party's telephone equipment or the content of recording is stored in the parameter storage unit 31 as a parameter, and the control unit 32 does not use the amount of attenuation appropriately read from the parameter storage unit 31 during recording. The audio period attenuation processing unit 29 may be set.

  By the way, the instantaneous power of a call signal or a mixing call signal is estimated, a period in which the instantaneous power estimate exceeds a predetermined threshold is determined as a voiced period in which sound exists, and the instantaneous power estimate is less than the threshold. A silence period detector that determines that a period is regarded as a silence period in which no sound is present, and when the silence period detector does not detect a silence period (during a sound period), the call signal or the mixing call signal is used as it is. A silence period attenuation processing unit that outputs to the recording / playback signal processing unit 21 and outputs to the recording / playback signal processing unit 21 after the speech signal or mixing call signal is attenuated when the silence period detection unit detects the silence period. These may be provided in the recording unit 20 instead of the silent period detecting unit 28 and the silent period attenuation processing unit 29, respectively. In this way, if the call signal or the mixing call signal in the silent period is attenuated, unnecessary sound is not recorded, and the sound can be reproduced more clearly during reproduction.

  Note that a means for removing a speech signal or a mixing speech signal during a silent period or a silent period from a signal recorded by the recording unit 20 is used in the recording unit 20 instead of the silent period attenuation processing unit 29 or the silent period attenuation processing unit. If provided, there is an advantage that the memory capacity required for recording in the data storage unit 31 can be reduced.

(Embodiment 3 )
As shown in FIG. 4, the base unit M of the present embodiment has the same basic configuration as that of the reference example . Therefore, the same components are denoted by the same reference numerals and description thereof is omitted.

  The present embodiment is characterized in that the recording unit 20 includes a noise canceller unit 40 that is a noise removing unit that removes a noise component that is regularly included in a call signal to be recorded or a mixing call signal. The noise canceller unit 40 is composed of, for example, a band-pass filter that passes only the frequency band of a human voice, and removes background noise having a frequency lower or higher than that of the human voice. However, the noise canceller unit 40 is realized by using a DSP common to the recording unit 20 and controlling the DSP hardware by software (program).

  Thus, in the present embodiment, the noise canceller unit 40 removes stationary noise components included in the call signal or the mixing call signal, so that unnecessary noise is not recorded, and as a result, the sound is reproduced clearly during playback. Is something that can be done.

It is a block diagram which shows the reference example of this invention . 1 is a block diagram illustrating a first embodiment. FIG. 6 is a block diagram illustrating a second embodiment. FIG. 6 is a block diagram illustrating a third embodiment.

Explanation of symbols

M Interphone main unit 1 Microphone 2 Speaker 4 Loud call processing unit 20 Recording unit 21 Recording / playback signal processing unit 22 First volume adjustment unit 23 Second volume adjustment unit 24 Mixing unit 25 Third volume adjustment unit 30 Data storage unit

Claims (7)

An intercom device that performs two-way loudspeaking calls through a communication path that is appropriately formed between a telephone device installed inside a house and outside the house, and transmitting a microphone, a speaker, and a sound collected by the microphone A voice switch that passes only one of a speech signal and a reception signal sent from the other party's telephone device, a first echo canceller provided between the voice switch and the microphone and speaker, and provided between the voice switch and the telephone device A voice call processing means for realizing a two-way voice call with the second echo canceller provided, a transmission signal output from the first echo canceller, and a reception signal output from the second echo canceller are recorded. and recording means for recording a signal mixing the transmission signal and the received signal together, to adjust the sound volume of the reception signal sent to the recording unit A first volume control means, and a second volume control means for adjusting the volume of the transmission signals sent to the recording unit, a third sound volume adjustment transmission signal and the reception signal to adjust the volume of the mixed signal Means, fourth volume adjusting means for adjusting the volume of the transmitted signal before mixing, fifth volume adjusting means for adjusting the volume of the received signal before mixing, and a transmitted signal for each of a plurality of types of telephone equipment And a volume adjustment value storage means storing at least one volume adjustment value of the received signal, and a volume adjustment value corresponding to the type of the other party's call device when recording by the recording means is read from the volume adjustment value storage means An interphone device comprising volume adjustment value setting means for setting to first to fifth volume adjustment means . A silent period detecting means for detecting a silent period in which there is no sound of the signal recorded by the recording means, and a silent period for attenuating the signal recorded by the recording means during the silent period detected by the silent period detecting means The intercom device according to claim 1 , further comprising an attenuation unit . A silence period detecting means for detecting a silence period in which there is no sound of a signal recorded by the recording means, and a silence period attenuating means for attenuating a signal recorded by the recording means during the silence period detected by the silence period detecting means. The intercom device according to claim 1, wherein 4. The interphone device according to claim 2 , wherein the silent period attenuating means or the silent period detecting means adjusts the amount of attenuation according to the type of the counterpart telephone equipment. A silent period detecting means for detecting a silent period in which there is no sound of the signal recorded by the recording means, and a removing means for removing the silent period detected by the silent period detecting means from the signal recorded by the recording means; The intercom device according to claim 1, further comprising: A silence period detecting means for detecting a silence period that can be regarded as having no sound of a signal recorded by the recording means, and a removing means for removing the silence period detected by the silence period detecting means from the signal recorded by the recording means The intercom device according to claim 1 . Intercom device according to claim 1, further comprising a noise removal means for removing noise components contained in the stationary to the signal to be recorded in recording means.

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