JP2001024746A - Telephone set incorporating conference speech function - Google Patents

Abstract

PROBLEM TO BE SOLVED: To obtain a telephone set capable of satisfactorily realizing a conference function by having a means for selecting a specific person from plural speaking opposite parties in a conference speech, to adjust a receiving speech volume to secure the balance of the receiving speech volume between with each speaking opposite party. SOLUTION: A telephone set has a means for selecting a specific person from plural speaking opposite parties in a conference speech to adjust a receiving speech volume to secure the balance of the receiving speech volume between with each speaking opposite party. In this telephone set, a multiplex timing generating part 16 decodes the number of conference attendants designated from a central processing part 19 to generate a multiplex timing signal for time-division-multiplexing voice data for the number of speaking opposite parties. A volume control part 17 stores a relative volume value designated from the part 19 in a buffer part 2 or 3 corresponding to the speaking opposite party in advance and reports it to a digital volume par 5 by synchronizing it with a time-divisionmultiplexing timing. The part 5 decodes relative volume value from the part 17 to execute digital volume arithmetic.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

The present invention relates to a telephone having a conference call function for realizing a conference function.

[0002]

2. Description of the Related Art A conference call system in an exchange is known, but a conference call system in a conventional exchange has a voice mixing function for holding a conference on a communication path of the exchange. It can be said that the switching function of the voice communication path is entirely performed by the exchange, and is inevitable. Hereinafter, this method is defined as a centralized type.

FIG. 7 shows the concept of the centralized conference call system.
The operation of the centralized conference call system will be described with reference to FIG. The telephone set A, the telephone set B, the telephone set C, and the exchange body EX shown in FIG. 1 are conventional exchange systems typified by ordinary PBXs and key telephones. A conference call system in this configuration will be described.

[0004] First, the structure of the normal call system in this configuration will be described. Telephones A, telephones B, and telephones C are transmitted to the exchange body. In the exchange body, the connection destination is determined by a communication path that determines the connection destination of each transmission voice. For example, for a telephone call of A: telephone and a telephone: B, the transmission voice of telephone: A is connected to the reception of telephone: B. Then, the transmission voice of the telephone set B is connected to the reception of the telephone set A, so that the communication state is established.

[0005] Next, the mechanism of the conference call system in this configuration will be described. In this case, the exchange body EX manages the conference path and performs audio mixing. Here, a case where the telephone set A, the telephone set B, and the telephone set C perform a three-party call will be described. In the case of the conference call system, the transmission of the transmission voice is the same as that of the normal call system, but the voice connected to the reception is different from the normal call system, and each of the telephone: A, telephone: B, and telephone: C is transmitted. Speech voices (arrows A, B, and C from the telephones to the exchange main body in the figure) are mixed by A + B, A + C, and B + C in the exchange main body, respectively, and then mixed voices of A + B (in the figure, telephones from the exchange main body to C: C). Arrow A + B) toward the telephone: to listen to C,
A + C mixing voice (in the figure, an arrow A + C from the exchange body toward the telephone: B) is transmitted to the telephone
+ C mixing voice (in the figure, from the exchange itself to the telephone:
A conference call is established by connecting the arrow B + C) toward A to the telephone: A's receiver.

[0006]

The mechanism of the centralized conference call system has been described above. In this case, the following points can be pointed out in consideration of the following aspects.

Problem (1): Reception volume control (a) Volume control in the centralized conference call system is performed by a switch body or a terminal such as a telephone, and the reception volume is varied irrespective of a normal call or a conference call. That is what it was doing. FIG. 8 shows the concept (a) of volume control by the exchange body in the centralized conference call system.

(B) FIG. 1 shows an example in which the exchange body has a volume control function in the centralized conference call system.
In this example, a telephone terminal unit composed of telephones: A to C and a LINE card: A to C, a mixing unit of A + B, A + C, B + C, and a communication path unit composed of a reception volume unit of VOL # 1 to # 3. Consists of In this case, the reception volume units VOL # 1 to # 3 are all mixed (A + B, A + C, B + C in the preceding mixing units, respectively).
After the mixing voice is processed). As a result, the following state may be caused.

(C) For example, in the telephone: A, B +
Assume that while listening to the C mixing sound, the B sound is at a low volume and the C sound is at an optimum volume for the user of the telephone: A. At this time, since the B sound has a small volume, if the B sound is adjusted to an optimal volume, in this configuration, the C sound becomes large to control the volume of the B + C mixing sound, and the volume cannot be balanced. ). This problem (problem) is the same even when the telephone has a volume control function, since the control of the volume of the mixed voice remains unchanged.

(D) Then, the following countermeasures can be considered. FIG. 9 shows a concept (b) of volume control by the exchange body in the centralized conference call system as a countermeasure against the above-mentioned problem (comparative example). In the present configuration, the volume control is performed before the mixing, thereby controlling the volume of each voice individually, thereby clearing the problem in the case of FIG.

(E) However, in the present configuration, the number of VOLs is twice as many as that in FIG. 8 for a three-party call (the volume is VOL # 1 to VOL # 6 shown in FIG. 9 for a total of six VOLs). necessary)
There are problems (problems) such as an increase of three times in a four-party call and a four-fold increase in a five-party call, and complicated processing of the exchange body.

[0012] Therefore, it is desirable to be able to balance the reception volume without any disadvantages or inconveniences such as not being able to balance the volume as described above in a conference call with a plurality of communication partners. Therefore, it is possible to secure a receiving sound volume balance with each of the other parties, and more desirably, the number of necessary volume parts does not increase as in the comparative example, and the number is as small as possible, a simple configuration, and the like. Can be realized.

Problem (2): Speaker display (g) Next, what can be improved is the speaker display. In a conference call with a plurality of communication partners, who is the speaker? If it can be known, the conference by calling can be made smoother and more efficient in terms of content and progress.

(H) The telephone reception in the centralized conference call system is a mixed voice. For this reason,
It has not been possible to display who is the speaker on a telephone. This is because the mixed sound cannot be restored unless the original sound is known. For example, in order to restore the voice A from the A + B mixed voice, the voice B is required, but the switch body in the centralized conference call system cannot transmit individual voices to the telephone (FIG. 7, FIG. 7). 8). This is the same for the comparative example (FIG. 9).

[0015] Therefore, it is desirable that, in a conference call having a plurality of parties, it is possible to display the speaker who is speaking. That is, it is possible to display the name of the person who is currently speaking from the other party, the telephone number of the other party, and the like.

The present invention can solve the above-mentioned disadvantages and inconveniences based on the above-mentioned considerations and the following considerations, and can realize a good conference function. The aim is to provide a telephone with a built-in conference call function.

[0017]

According to the present invention, the following telephone having a conference call function is provided. That is, the present invention is a telephone with a built-in conference call function, which has means for selecting a specific person from among a plurality of parties in a conference call, adjusting the reception volume, and securing the reception volume balance with each party. It is characterized by the following. Further, in the conference call, the communication device has a communication partner designation input means for designating a communication partner.

The present invention is characterized in that it has a time-division multiplexing unit and a demultiplexing unit for voice processing in a conference call, and is constituted by one component block group necessary for mixing. Also, in a conference call, a function of changing a multiplex timing according to the number of callers is provided. In a conference call, a function of notifying a volume unit of a plurality of volume values for each of the other parties in synchronization with multiplex timing is provided.

In the case of a normal call, the volume of voice data bypassed by the bypass unit is controlled, and the volume for the conference call and the volume for the normal call are shared. .

[0020] The present invention is also characterized in that it has a function of detecting the voice level of each communication partner. In addition, there is provided a means for executing voice level detection, specifying a speaker from a plurality of communication partners, and visually displaying the speaker. Also, the means for visually displaying the speaker is an LED, and is associated with the other party designation input means for designating the other party, so that the state of the corresponding LED is activated and the other party designation input means is activated. Thus, it has a function of adjusting the volume of the sound. Further, the present invention is characterized in that the means for visually displaying a speaker is a display device, and has a function of displaying at least the name and / or telephone number of the speaker by specifying the speaker. .

[0021]

The present invention has the following advantages and features. [1] According to the present invention, in a conference call in which a plurality of callers are present, it is possible to pick up one of the callees and adjust the reception volume, thereby enabling the reception volume to be balanced. There is. [2] According to the present invention, volume control can be performed by multiplexing audio data by the time division multiplexing unit. Therefore, in the case of the system employing time division multiplexing, one digital volume is used. There is an effect that can be constituted only by. [3] According to the present invention, since the audio data is multiplexed by the time division multiplexing unit, there is an effect that, for example, only one PCM-linear conversion unit can be configured. [4] According to the present invention, in the case of a normal call, the volume of audio data bypassed by the bypass unit can be controlled, and the volume for a conference call and the volume for a normal call can be shared. . [5] According to the present invention, by having a voice level detection unit and a speaker display unit, picking up a voice currently being spoken among a plurality of communication partners and displaying a speaker. By using an LED or a display for the speaker display section, it is possible to visually represent the situation of a call. [6] According to the present invention, when a display is used for the speaker display unit, the name of the person who is currently talking is specified from among a plurality of parties, and the name and the other party are displayed on the display. There is an effect that a telephone number or the like can be displayed.

The telephone having a conference call function according to the present invention can be applied to and used in all systems employing a distributed conference call system, such as a LAN telephone system, capable of executing conference processing on a terminal side. .

The above and other objects, advantages, features and the like according to the present invention will become more apparent from the following description.

[0024]

Embodiments of the present invention will be described below with reference to the drawings. FIG. 1 is a diagram (functional block diagram) showing a configuration of an embodiment of a telephone having a conference call function according to the present invention.
It is also a principle diagram of the present invention (telephone with a built-in conference call function in the case of adopting time division multiplexing). Also,
FIG. 2 is a diagram showing the concept of a distributed conference call system used for explaining the present invention.

Here, prior to the description of the configuration of FIG.
First, the basic concept and features (or necessary prerequisites) adopted by the present invention to solve the above-described problems will be described with reference to FIG. 2, and then further described with reference to FIG. A detailed description will be added. First,
In the present invention, the precondition (basic concept) is based on a distributed conference call system.

The distributed conference call system will be described with reference to the concept of the distributed conference call system shown in FIG. The feature of this configuration is that, as shown in the figure, there is no communication path in the call management device TS, and the role of the call management device TS is to transmit the voice to be received to each telephone (A, B, C in the illustrated example). In this case, each telephone sets a telephone call state by taking in the received voice based on the timing of the voice information received and the header information based on the notification.

[ Case of Normal Call ] The mechanism of the normal call method in this configuration will be described. Phone: A, Phone:
B, telephone: C Each transmitted voice (in the figure, arrow A from telephone A to telephone B, telephone C), arrow B from telephone B to telephone A, telephone B to arrow C, telephone: C Arrows C) directed from C to the telephone set A to the telephone set B are transmitted in chronological order in a synchronized system, or transmitted at an arbitrary timing as a packet with a header added like a LAN. . In order to connect these transmission voices to the receiving side, under the control of the call management device TS, the call management device TS
In response to the notification of the timing of the voice to be received and the notification of the header information from each of the telephones, each telephone takes in the transmission voice, thereby bringing the telephone into a call state.

For example, in the case of a telephone call between the telephone set A and the telephone set B, the telephone set B captures the transmission voice of the telephone set A and the telephone set A captures the transmission voice of the telephone set B. In the state.

[ Case of Conference Call] Next, the mechanism of the conference call system in this configuration will be described. Here is the phone:
A case will be described where A, telephone: B, and telephone: C make a three-party call. In the case of the conference call system, the transmission of the transmission voice is the same as in the normal communication system, and the reception of the reception voice is also the same as in the normal communication system. The difference here is that the incoming voice is captured by telephone: A in telephone:
The transmission voice of B and C, the transmission voice of telephones A and C in telephone: B, and the voice of telephones A and B in telephone: C are respectively taken in, and the receiving side mixes them into a conference call. That is. The call management device TS manages the conference path.

The outline of the distributed conference call system has been described above. As described above, the distributed type is not a centralized type using a switch such as a conventional switching system, but also performs a mixing process of a conference and a voice call process such as a reception volume in a terminal such as a telephone in a distributed manner. is there. The present invention is based on this concept.

Referring to FIG. 1, there is shown an example of a configuration according to the present invention which can perform such conference mixing processing and voice communication processing such as reception volume. In the figure, N
T indicates a network, and an audio data reception processing unit 1 for executing a reception process of audio data received from the network is provided for T. In addition to the audio data reception processing unit 1, in the case of the illustrated example, the buffer units 2, 3,
Time division multiplexing unit 4, digital volume unit 5, PCM
A linear conversion unit 6, a separation unit 7, a digital mixing unit 8, a linear-PCM conversion unit 9, a parallel / serial conversion unit 10, a CODEC unit 11, an amplifier unit 12, a receiver and a speaker 13, and bypass units 14 and 15; ,further,
Multiplex timing generator 16, volume controller 17, audio level detector 18, central processor 19, speaker display 2
0, there is a man-machine interface 21.

Here, the buffer units 2 and 3 are buffer units for storing the audio data (audio data: A, audio data: B in the illustrated example) received by the audio data reception processing unit 1 (audio data: B). : Buffer section for A and audio data: buffer section for B). Therefore, in the context of FIG. 2, the configuration illustrated in FIG. 1 corresponds to the telephone: C which takes in the voices from the telephone: A and the telephone: C in the distributed conference call system of FIG. This corresponds to the case of

The time division multiplexing section 4 is a time division multiplexing section for time division multiplexing each audio data. The digital volume unit 5 is a digital volume unit for controlling a received voice volume. The PCM-to-linear converter 6 is a PCM-to-linear converter that is executed as a preparation before executing the digital mixing. The separation unit 7 is a separation unit for separating time-division multiplexed audio data. The digital mixing unit 8 is a digital mixing unit that performs a conference call function. In the present embodiment, the time-division multiplexing unit 4 and the demultiplexing unit 7 are provided in order to increase the efficiency of voice processing in a conference call,
The configuration block group necessary for mixing does not require the same number of blocks as the number of the other parties (in the example of FIG. 2, the two parties, which are the number of the other parties of the telephone: A and the telephone: C). (Without need).

In this case, in this embodiment, the audio data is multiplexed by the time division multiplexing unit 4 to control the volume (digital volume unit 5-volume control unit 17-central processing unit 19-man-machine interface). As shown in the figure, the digital volume to be used can be constituted by only one digital volume. In addition, since the time-division multiplexing unit 4 multiplexes audio data, the PCM-to-linear conversion unit 6 may be configured with only one PCM as shown in the figure.

The linear-PCM conversion unit 9 is a linear-PCM conversion unit that converts linear audio data into PCM data. The parallel-to-serial converter 10 is a parallel-to-serial converter that converts parallel data into serial data. The CODEC unit 11 is a codec unit that converts PCM data into an analog signal. The bypass units 14 and 15 are used as bypass units for bypassing the conference processing unit in the case of the normal call described above.
Here, in the case of a normal call, the volume of the audio data bypassed by the bypass unit is controlled (digital volume unit 5-volume control unit 17-central processing unit 19-man-machine interface unit 21, etc.). Therefore, a configuration is possible in which the volume for conference calls and the volume for normal calls are shared by one.

The multiplex timing generator 16 is a multiplex timing generator that generates time division multiplex timing.
The volume control unit 17 is a volume control unit that controls a volume value for a digital volume.
The voice level detecting unit 18 is a voice level detecting unit that detects a voice level of a speaker on the speaker display unit 20. The function of detecting the voice level of each call partner includes the voice level detection unit 18 and realizes this. The central processing unit 19 is a central processing unit that controls various functions and controls the entire operation. The man-machine interface unit 21 is a man-machine interface for controlling a volume.
An interface unit can be configured.

In the case of the telephone set A and the telephone set C shown in FIG.

Among the above-mentioned units, the processing block that is the main backbone (point) of the telephone with a built-in conference call function according to the present invention is based on the time-division multiplexing unit 4 with respect to the aforementioned problem (1) “Received sound volume control”. , Digital volume unit 5,
The bypass unit 14, the multiplex timing generation unit 16, the volume control unit 17, the central processing unit 19, the man-machine interface unit 21, and the like. A level detector 18 and a speaker display 20 are included. This basically provides a function of selecting a specific person from a plurality of parties in a conference call, adjusting the receiving volume, and ensuring a receiving volume balance with each party. First, an outline of the operation of the processing block which is the point will be described.

The operation of the man-machine interface unit 21 is an operation content for controlling an interface (connection function) between a user (conferee) and a device (telephone with a built-in conference call function used by the conferee). is there. This feature will be described later in FIG. 4 (man-machine interface of a telephone panel), but it has buttons for the number of conference participants, and after pressing the corresponding button, some (predetermined) service button is pressed. By pressing this button, various services can be provided for each received voice. Here, the “button” refers to all input means, such as a touch panel and a mouse, which are widely used as a man-machine interface.

The central processing unit 19 processes information from the man-machine interface unit 21 and notifies the multiplex timing generation unit 16 of the number of conference callers, the volume control unit 17 of the volume value notification, The information from the level detector 18 is processed, and the speaker display unit 20 is notified of speaker information. In this case,
Under the control of the central processing unit 19, as described in FIG. 3 (an example of a time chart of multiplex timing) described later, in the conference call, a process for changing the multiplex timing according to the number of communication parties can be included. The process of notifying the volume value can include, in a conference call, a process of notifying the volume unit of the volume value of each of a plurality of communicating parties in synchronization with the multiplex timing.

The multiplex timing generation section 16 decodes the number of conferees specified by the central processing section 19, and generates a multiplex timing signal for time-division multiplexing audio data for the number of communication parties.

The time-division multiplexing unit 4 receives the multiplexed timing signal from the multiplexed timing generating unit 16 and, in the present principle diagram, stores the audio data: A buffer (2) and audio data: B buffer (3). Each audio data is time-division multiplexed. In principle, the number of buffers may be two or more (even in such a case, one component block group including the volume part (5) and necessary for mixing is sufficient as described above). The time-division limitation depends on the clock speed of the time-division clock.

The bypass unit 14 performs a normal call (receiving: 1).
Time) multiplexing timing generator 16 to time-division multiplexer 4
Is instructed to be bypassed.

The volume control unit 17 includes a central processing unit 19
Is stored in a buffer (buffer unit 2 or 3 in this example) corresponding to the communication partner, and is notified to the digital volume unit 5 in synchronization with the time division multiplex timing. The digital volume unit 5 decodes the relative volume value from the volume control unit 17 and executes a digital volume operation.

By the above-described processing of each part, it is possible to control the receiving volume of each party with respect to the points to be considered ((a) to (f)) in the first problem (1). The receiving volume can be optimized at the same time, and as described in the points (a), (b), and (c) of the problem (1), there is a disadvantage that the volume of each party cannot be balanced during the conference. This can be cleared without causing any problems (problems). Moreover, in the case of the present embodiment, even when compared with the comparative example (FIG. 9), this is realized without causing the disadvantages (problems) pointed out in the above-mentioned consideration points (d) and (e). It is possible to Hereinafter, with reference to FIG. 3 and subsequent figures, the operation of each component for solving the problem (1) will be described in detail.

First, the transition from a normal call to a conference call will be described. In the case of a normal call (the other party: one person),
A central processing unit 19 to a multiplex timing generation unit 18;
The volume control unit 17 is notified that the call is a normal call. Upon receiving this, the multiplex timing generation section 18 activates the bypass sections 14 and 15 to thereby activate the time division multiplex section 4, the PCM-linear conversion section 6, the separation section 7, the digital mixing section 8, and the linear-PCM section. Converter 9
Is established, a data transfer route for audio data is established by transferring the data to the
The voice is heard from the receiver 13 via the DEC unit 11.

The transition from the normal call state to the conference call state is performed according to the concept of the distributed conference call system shown in FIG.
Is started by analyzing.

The central processing unit 19 must recognize the number of callers participating in the conference included in the instruction of the conference call, and construct a voice path route according to the number of callers. The operation of the central processing unit 19 first recognizes the number of communicating parties, and secures the number of voice data buffers for the channels based on the number. In FIG. 1, a three-party call (a three-party call between a conferee using a telephone: A, a conferee using a telephone: B, and a conferee using a telephone: C) is an example. As described above, the number of buffers is two (audio data: buffer unit 2 for A and audio data: buffer unit 3 for B).

Next, the multiplex timing generator 16 is notified of the number of callers, and the volume controller 17 is notified of the volume value of each caller.

The multiplex timing generator 16 which has received the notification
Generates the timing signal shown in FIG. This time signal is notified to the time division multiplexing unit 4, and the time division multiplexing unit 4 receiving the multiplex timing signal
Audio data: A buffer unit 2 and audio data: B
Time-division multiplexing is performed by reading audio data from the buffer unit 3 at a timing according to the multiplex timing signal.

The volume control unit 17 is notified of the volume value corresponding to each channel of each communication partner, and synchronizes the notified volume value with each time slot (TS0 to TSn) as shown in FIG. The digital volume unit 5 is notified in such a form.

Here, as a supplementary explanation, FIG. 3 shows that an audio signal is basically 8 KHz sampling, and in the case where another channel is used, a time signal is used.
It needs to be processed in a time-sharing manner with the concept of slots. The number of channels is determined by the serial transfer clock rate. For example, for a clock of 2.048 MHz,
2.048 MHz / (8 kHz × 8 bits) = 32 channels.

The notations “A”, “B”,.
As for "A" and "B", the principle diagram of FIG.
And audio data: indicates that the A buffer unit 3 is to be subjected to time division multiplexing. Multiplex timing generator 16
Receives the notification of the number of callers from the central processing unit 19 and transmits the timing signal of FIG. The time division multiplexing unit 4 reads the audio data from each buffer at this timing and executes time division multiplexing.

Further, the notations “0 dB” and “0d” in FIG.
B ”...“ +3 dB ”,“ 0 dB ”, etc., means that the central processing unit 19 notifies the volume control unit 17 of the relative volume value for each channel. Upon receiving this, the volume control unit 17 notifies the relative volume value to the digital volume unit in synchronization with the multiplex timing signal generated by the multiplex timing generation unit 16. Further, in the example of this figure (the example of FIG. 3), the following reception sound volume control scene is illustrated. That is, this is a case where the telephone user (the telephone: the conferee using the telephone C) makes a volume increase request via the man-machine interface unit 21 because the receiving volume of the other party: A is low. After processing, 8KHz sampling timing. 1 and 8 KHz sampling timing. 2 is an example in which the reception relative volume value of the communication partner: A is increased (from 0 dB to +3 dB). Further, in this case, it will be understood that the reception volume of the other party B is not unnecessarily increased (the relative volume value of the other party B is 0).
dB). As a result, the receiving sound volume balance is appropriately secured.

Returning to FIG. 1, the notified digital volume unit 5 sets the relative volume value (for example, 0d
B, +3 dB, etc.), digital volume operation is performed on the audio data input for each time slot from the time-division multiplexing unit 4, and the calculated audio is transmitted to the PCM-linear conversion unit 6. Send data.

The PCM-to-linear converter 6 converts the calculated audio data (PCM) to linear data, and executes preparations for digital mixing. The separation unit 7 separates the time-division multiplexed linear audio data and transmits it to the digital mixing unit 8 in the same phase.

The digital mixing section 8 performs a mixing process on each of the separated linear audio data. In the linear-PCM conversion unit 9, the CODEC unit 1
1 converts digital mixed linear audio data into PCM data for signal processing.

The subsequent processing is the same as in a normal call, whereby a data transfer route for voice data is established by transferring the data to the parallel / serial conversion unit 10, and the mixed voice is heard from the receiver 13 via the CODEC unit 11. .

Thus, according to the present embodiment, in a conference call having a plurality of communication partners: A and B, one of the communication partners (in FIG. 3, the communication partner: A) is picked up and the reception volume is adjusted. Can be adjusted,
It is possible to balance the reception volume. The same applies to the telephone A in FIG. 2 (in this case, the other party in the conference call is the other party B, C).
It goes without saying that the same operation can be performed on the telephone B side (in this case, the other side of the conference call is the other side: A and C) in accordance with the above.

In a conference call, the other party designation input means for designating the other party can be provided in the man-machine interface section 21 of each telephone set according to the present embodiment. The man-machine interface unit 21 allocates a call partner designation input unit for each call partner, activates the input unit, and then sets the volume control input unit as shown in the example of the board of the telephone 100 illustrated in FIG. By making it active, the volume of each call partner can be changed. This function is a trigger for executing the receiving sound volume control for each call partner.

Here, in the example of FIG. 4, the telephone (telephone with a built-in conference call function) 100 is a dial unit 11.
0, a handset 120, a function button section 130,
A display unit 140 (for example, an LCD) and a function button (key) unit 130 are provided with call partner designation input means (A, B) 131-n (n =
, And an operation unit (up, down) as the volume control input unit 132 are included. As described above, the input means of the man-machine interface unit 21 shown in the principle diagram of FIG. 1 can be realized and implemented by keys (buttons).

The above is the description of the principle of controlling the receiving sound volume of each communication partner at the time of a conference and the corresponding components in the present embodiment, which are adopted for solving the problem (1).

Next, with regard to points ((g) to (g)) to be considered in speaker display of the problem (2), details of the operation of each component for solving the problem will be described. In the case of FIG. 1, the telephone 100 according to the present invention executes voice level detection by the system of the voice level detection unit 18-central processing unit 19-speaker display unit 20 included therein, and a plurality of communication partners The speaker is identified from among, and the speaker is visually displayed. In this case, preferably, the means for visually displaying the speaker is, for example, an LED, and is associated with the above-mentioned call partner designation input means 131-n, so that the user can provide the call partner designation input means in advance. A configuration is provided in which the function of adjusting the volume of the sound is provided by activating the other party's designation input means by looking at the state of the corresponding LED without setting the other party, and preferably, The means for visually displaying is a display device, and by specifying a speaker, a function of displaying the name and telephone number of the speaker can be provided.

For this reason, the audio level detector 18 averages the absolute audio levels of the sampled audio data time-divided for each communication partner after executing the digital volume processing, and determines each audio level after the audio level is determined. The level is notified to the central processing unit 19 for each time slot. Alternatively, the audio level detection unit 18 notifies the central processing unit 19 of the absolute audio level of each sampled audio data for each time slot, and the central processing unit 19 averages the absolute audio levels of each sampled audio data. Good. The averaging here means averaging the absolute level indicated for each sampling cycle with an arbitrary number of samples to obtain a sound level. An arbitrary number of samples is, for example, when detecting a sound level in one second, from the sampling cycle (8 kHz), 1 s / 125 us = 8,000
Be a sample. The averaged audio level for each time slot is processed in the central processing unit 19, and the speaker can be displayed by controlling the speaker display unit 20.

Here, the speaker display section 20 is intended for all objects visually expressed (visual display means). Typical examples thereof include an LED and a display. The central processing unit 19 manages the telephone number of the other party and the name of the other party corresponding to the telephone number. If this information is notified to the speaker display unit, it is possible to identify the person who is currently talking and display the speaker on the speaker display unit even when there are a plurality of parties during a conference call. . Thus, according to the present embodiment, in addition to the effects described above, the presence of the voice level detection unit 18 and the speaker display unit 20 allows the voice currently being spoken among a plurality of communication partners to be present. Can be picked up and a speaker can be displayed, and by using an LED or a display for the speaker display section 20, the situation of a call can be visually expressed. In addition, when a display is used for the speaker display unit 20, the name of the person who is currently speaking is specified from a plurality of parties, and the name, the telephone number of the other party, and the like are displayed on the display. There is an effect that can be displayed.

Further, L is input to each of the other party designation input means 131-n (n = 1, 2,...) In the example of the telephone board shown in FIG.
The person who is currently talking is specified by associating the ED with the LED by blinking or lighting or by color identification. For example, when the reception volume of the other party is low, the corresponding party designation input means corresponding to the LED After activating 131-n, by activating the volume control input means 132, it is possible to control the received sound volume for each call partner as described in the above problem (1). Therefore, in this case, the user activates the communication partner designation input means 131-n by checking the state of the corresponding LED without setting the communication partner designation input means 131-n in advance. Thus, the volume of the sound can be adjusted. The present invention can also be implemented in this way.

In implementing the telephone having the conference call function of the present invention, it has already been described that the input means of the man-machine interface unit 21 in FIG. 1 can be realized by a key (button) of the telephone. The unit (20) can be realized by an LED and / or a display. If the display (20) is used, the display unit 140 (FIG. 4) of the telephone 100 can be used. FIG. 1, which also has a principle as a principle diagram, describes a three-party call (two-party reception), and describes a voice data buffer unit (2, 3).
Is based on the number of time slots of multiplex timing, although only two are shown in the figure.
If two time slots are targeted, the number of audio data buffer units is 32. The processing of the central processing unit 19 can be performed by either software or hardware.

Next, another embodiment of the present invention will be described with reference to FIGS. FIG. 5 shows another embodiment of the present invention, which can be regarded as a modification or modification of the above-described embodiment (first embodiment). As shown in FIG. 5, the parts related to the time-division multiplexing unit 4, digital volume unit 5, PCM-linear conversion unit 6, demultiplexing unit 7, bypass unit 14, and multiplex timing generation unit 16 in the first embodiment are the same as those in the present embodiment. In the second embodiment, the configuration is changed as shown in the figure and has digital volume units 31 and 32 and PCM-linear conversion units 33 and 34, and can be implemented without using time division multiplexing. ing. The operation principle, operation and effect of each of the other constituent blocks are the same as those in FIG. The present invention can also be implemented in this way.

In this modified example (first modified example), the time division multiplexing is stopped, so that the time division multiplexing section 4, the separation section 7,
Since there is no bypass section 14 and no multiplex timing generation section 16, as shown in FIG.
1, 32 and the PCM-to-linear converters 33 and 34, the digital volume unit and the PCM-to-linear converter are required for channels that support conference calls.

FIG. 6 shows still another embodiment of the present invention, which can be regarded as a modified example or a modified example of the first embodiment, and a modified example of the second embodiment. It can also be seen as a modification. As shown in FIG. 6, the time division multiplexing unit 4, digital volume unit 5, PCM in the first embodiment
A linear conversion unit 6, a separation unit 7, a digital mixing unit 8, a linear-PCM conversion unit 9, a parallel / serial conversion unit 10, a CODEC unit 11, bypass units 14, 15,
In the present example (third embodiment), the parts related to the multiplex timing generation unit 16 are changed as shown in the figure, and the serial / parallel conversion units 41 and 42, the CODEC units 43 and 44, the analog volume units 45 and 46, and the analog -It has the structure which has the mixing part 47.

A feature of this modified example (second modified example) is that the mixing process is performed in analog. By performing the mixing in analog, there is no digital mixing processing part, and the audio data is sent directly from the audio data buffers (2, 3) to the CODEC units 43, 44 through the serial / parallel conversion units 41, 42. After mixing is performed by the analog mixing unit 47 via the analog volume units 45 and 46,
Listening from the receiver 13. The present invention can also be implemented in this way.

In this modification, the serial-to-parallel converters (41, 4) for the number of channels supporting conference calls are used.
2), a CODEC section (43, 44) and an analog volume section (45, 46) are required, and the audio level detection section 18 needs to have an analog / digital conversion circuit.

In any of these modifications, similarly to the first embodiment, a telephone with a built-in conference call function is applied to each of the telephones: A, B, and C in FIG. It can be used for

The present invention is not limited to the above embodiment. For example, the contents described in the above-described embodiments and modified examples can be regarded as the following inventions.

[Appendix 1] A telephone with a built-in conference call function having a function of selecting a specific person from a plurality of parties in a conference call, adjusting the reception volume, and securing the reception volume balance with each party. Or a distributed conference call system in which the telephone is used as a terminal to execute conference processing.

[Additional Item 2] In a conference call, a telephone with a built-in conference call function having a call partner designation input means for designating a call partner, or a distributed conference call system in which the telephone is used as a terminal to execute a conference process.

[Appendix 3] A time-division multiplexing unit and a demultiplexing unit are provided to increase the efficiency of voice processing in a conference call. A telephone with a built-in conference call function composed of one or a distributed conference call system in which the telephone is used as a terminal to execute conference processing.

[Appendix 4] A telephone with a built-in conference call function having a function (means) for changing the multiplex timing according to the number of parties in a conference call, or a distributed conference call system in which the telephone is used as a terminal to execute conference processing.

[Appendix 5] In a conference call, a telephone with a built-in conference call function having a function (means) of notifying the volume unit of a plurality of existing call partners in synchronization with the multiplex timing to the volume unit, or the telephone as a terminal A decentralized conference call system that executes conference processing.

[Supplementary Note 6] A telephone with a built-in conference call function having a function (means) of detecting the voice level of each call partner, or a distributed conference call system in which the telephone is used as a terminal to execute conference processing.

[Appendix 7] The voice level is detected, and
A telephone with a built-in conference call function having a function (means) for specifying a speaker from a plurality of existing parties and visually displaying the speaker, or a distributed conference call using the telephone as a terminal to execute conference processing method.

[Supplementary Item 8] The means for visually displaying the speaker is an LED, and the user sets in advance the other party in the other party designation input means by making it correspond to the above-mentioned other party designation input means. Without looking at the status of the corresponding LED and activating the call partner designation input means,
A telephone with a built-in conference call function having a function (means) for adjusting the volume of the voice, or a distributed conference call system in which the telephone is used as a terminal to execute conference processing.

[Appendix 9] The means for visually displaying the speaker is a display unit, and by specifying the speaker,
A telephone with a built-in conference call function having a function (means) for displaying the name and telephone number of the speaker, or a distributed conference call system in which the telephone is used as a terminal to execute conference processing.

[Brief description of the drawings]

FIG. 1 is a diagram for explaining an embodiment of the present invention, and also a principle diagram for explaining the principle thereof.

FIG. 2 is a diagram showing the concept of a distributed conference call system for explaining the present invention.

FIG. 3 is also a diagram showing an example of multiplex timing.

FIG. 4 is also a diagram showing a man-machine interface (an example of a telephone panel).

FIG. 5 is a diagram which is used for describing another embodiment (first modification) of the present invention.

FIG. 6 shows still another embodiment of the present invention (second modification).
FIG.

FIG. 7 is a diagram showing the concept of a centralized conference call system.

FIG. 8 is an explanatory diagram of the concept (a) of volume control by the exchange body in the centralized conference call system.

FIG. 9 is an explanatory view showing another concept (b) of another volume control by the exchange body in the centralized conference call system, shown as a comparative example.

[Explanation of symbols]

 Reference Signs List 1 audio data reception processing unit 2 buffer unit (audio data: buffer unit for A) 3 buffer unit (audio data: buffer unit for B) 4 time division multiplexing unit 5, 31, 32 digital volume unit 6, 33, 34 PCM -Linear conversion unit 7 Separation unit 8 Digital mixing unit 9 Linear-PCM conversion unit 10,41,42 Parallel / serial conversion unit 11,43,44 CODEC unit 12 Amplifier unit 13 Receiver (speaker) 14,15 Bypass unit 16 Multiplexing Timing generation unit 17 Volume control unit 18 Audio level detection unit 19 Central processing unit 20 Speaker display unit 21 Man-machine interface unit 45, 46 Analog volume unit 47 Analog mixing unit 100 Telephone (telephone with built-in conference call function) 110 dial part 120 handset 130 Function button unit 131 communication partner designation input means (button (key)) 132 Volume control input means (button (key)) 140 Display unit (display) A, B, C telephone TS speaking control unit NT Network

Continued on the front page (72) Inventor Kazuhiko Yoshida 6-12-1, Nishi-Shinjuku, Shinjuku-ku, Tokyo Fujitsu Eye Network Systems Co., Ltd. F-term (reference) 5K015 AA00 AB01 AB04 AF09 JA00 JA01 JA05 JA10 JA14 5K027 AA07 BB03 BB15 CC02 CC04 DD14 FF01 FF22 HH00 MM16

Claims (10)

[Claims] 1. A built-in conference call function, comprising means for selecting a specific person from among a plurality of parties in a conference call, adjusting a receiving volume, and securing a receiving volume balance with each party. Phone. 2. The telephone according to claim 1, further comprising: a telephone with a conference call function, wherein said telephone has a call partner designation input means for designating a call partner in a conference call. Phone with built-in conference call function. 3. A telephone with a built-in conference call function, comprising a time division multiplexing unit for voice processing in a conference call and a demultiplexing unit, and comprising a single group of constituent blocks required for mixing. Or claim 1 or claim 2
3. The telephone with a built-in conference call function according to claim 1, further comprising: 4. A telephone with a built-in conference call function, which has a function of changing multiplex timing according to the number of parties in a conference call, or further comprising the function. The telephone with a built-in conference call function according to any one of the above. 5. A telephone with a built-in conference call function, characterized in that it has a function of notifying a volume unit of a plurality of existing caller's volume values in synchronization with multiplex timing in a conference call, or further comprising the function. The telephone with a built-in conference call function according to any one of claims 1 to 4, wherein: 6. In a normal call, the volume of voice data bypassed by the bypass unit is controlled, and a volume for a conference call and a volume for a normal call are shared. 6. The telephone with a built-in conference call function according to claim 5. 7. A telephone with a built-in conference call function, which has a function of detecting a voice level of each call partner, or further comprising the detection function. A telephone with a built-in conference call function as described. 8. A telephone with a built-in conference call function, comprising means for executing voice level detection, specifying a speaker from a plurality of callers, and visually displaying the speaker. 8. The telephone with a built-in conference call function according to claim 1, further comprising said means. 9. A means for visually displaying a speaker is an LED, and by associating with the communication partner designation input means according to claim 2, the state of the corresponding LED is activated to activate the communication partner designation input means. 9. The telephone with a built-in conference call function according to claim 8, wherein the telephone has a function of adjusting the volume of the voice. 10. A means for visually displaying a speaker is a display, and has a function of displaying at least the name and / or telephone number of the speaker by specifying the speaker. The telephone with a built-in conference call function according to claim 8.