JP2005151042A - Sound source position specifying apparatus, and imaging apparatus and imaging method - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide an imaging apparatus and an imaging method capable of efficiently acquiring a clearer image with few imaging means, through the detection of the direction of a sound source. <P>SOLUTION: Each of two sound source direction detection sections 50a, 50b is provided with a plurality of microphones and detects the direction of the sound source, based on the sound pressure level of a sound signal from each microphone. A sound source position estimate section 70 geometrically estimates the sound source position in an imaging object chamber R1, on the basis of the direction of the sound source detected by the sound source direction detection sections 50a, 50b. Imaging sections 60a, 60b are controlled to carry out photographing, while being oriented to the estimated sound position. An image recognition section 80 applies image recognition processing to the photographed video data. The image recognition section 80 controls the zoom function of the imaging section 60a so as to display an object (sound source) in large size. <P>COPYRIGHT: (C)2005,JPO&NCIPI

Description

  The present invention relates to a sound source position specifying device, an image pickup device, and an image pickup method for picking up an image by specifying an object position based on sound information emitted from an object.

In recent years, an imaging apparatus that automatically captures an image with a predetermined space as an imaging target has been widely used.
For example, with the recent worsening of crimes, for safety (security), a camera as the imaging means of the imaging device is attached to a predetermined location in the monitored room, and the monitored room is photographed for a long time and recorded. To be done. In addition to security, for example, in order to monitor a child entrusted to a nursery school from the outside, the child's room is photographed by a surveillance camera installed in the room.
In such a case, conventionally, a monitoring camera is installed by selecting a place where the room to be monitored can be photographed in the widest possible range, and images obtained by the monitoring camera are recorded and managed.

By the way, in a conventional imaging apparatus, when one or a few cameras are installed in a room to be monitored as a surveillance camera, each camera must be always used at a wide angle so that the room can be photographed as widely as possible. However, there is a problem that it is difficult to obtain a clear image.
On the other hand, when a clear image is to be recorded over a wide range of a room to be monitored, a large number of monitoring cameras may be necessary, and there is a problem that the entire imaging apparatus is expensive.
Also, normally, no audio is recorded in the security video or the like, and when the video is reproduced later, there is no audio, so that the recorded information is not sufficient. For example, it is impossible to confirm how much damage has been received in the monitoring target based on voice.

  The present invention has been made in view of such circumstances, and an object of the present invention is to efficiently acquire a clear image with a small number of imaging means by specifying and capturing an object position based on sound information emitted from the object. An object of the present invention is to provide a sound source position identification device, an imaging device, and an imaging method.

  The first aspect of the present invention for achieving the above object is that at least two microphones having directivity are arranged so as to be directed in different sound collecting directions, and based on sound pressures collected by the microphones. Two or more microphone selection means for selecting one microphone and a plurality of microphones respectively selected by the two or more microphone selection means specify a sound source position based on directions directed by at least two microphones. A sound source position specifying device including sound source position specifying means.

  According to a second aspect of the present invention for achieving the above object, at least two microphones having directivity are arranged so as to be directed in different sound collecting directions, and based on sound pressures collected by the microphones. Two or more microphone selection means for selecting one microphone and a plurality of microphones respectively selected by the two or more microphone selection means specify a sound source position based on directions directed by at least two microphones. Sound source position specifying means, imaging means for changing the imaging direction by rotation, and imaging direction by rotating the imaging means so that the imaging means images the sound source position specified by the sound source position specifying means. And an imaging direction control means for controlling the imaging device.

  Preferably, the imaging apparatus further includes display means for displaying the video obtained by the imaging means, and enlarges and displays the sound source position specified by the sound source specifying position on the display means.

  Preferably, the imaging apparatus further includes image recognition means for recognizing a specific target from the video displayed by the display means, and enlarges the specific target recognized by the image recognition means and Display on the display means.

  According to a third aspect of the present invention for achieving the above object, at least two microphones having directivity are arranged so as to be directed in different sound collecting directions, and based on sound pressures collected by the microphones. A monitoring method comprising two or more microphone selection units for selecting one microphone and an imaging unit for changing an imaging direction by rotation, wherein a plurality of microphones respectively selected by the two or more microphone selection units is selected. Among these, the method includes a step of specifying a sound source position based on a direction in which at least two microphones are directed, and a step of rotating the imaging unit so that the imaging unit images the specified sound source position.

  According to the sound source position specifying device according to the first aspect of the present invention, at least two microphones having directivity are arranged in two or more microphone selection means so as to direct different sound collection directions. Each microphone selection means specifies a sound generation direction corresponding to the direction of the specified microphone in order to specify a microphone that determines that the collected sound signal from each microphone is a main sound generation based on the sound pressure level. The sound source position specifying means can uniquely determine the sound source position from the geometric relationship based on the direction in which at least two microphones are directed from the microphones specified by the two or more microphone selection means.

  According to the present invention, since a clear image can be efficiently acquired with a small number of imaging means, there is an advantage that the entire imaging apparatus is inexpensive.

Hereinafter, for convenience of description of first to third embodiments of the present invention to be described later, first, a communication device (bidirectional communication device) as a microphone selection unit of the imaging device of the present invention will be described.
FIGS. 1A to 1C are configuration diagrams showing an example to which the communication device of the present invention is applied.
As illustrated in FIG. 1A, communication devices 1A and 1B are installed in two remote conference rooms 901 and 902, respectively, and these communication devices 1A and 1B are connected by a telephone line 920. Yes.
As illustrated in FIG. 1B, in the two conference rooms 901 and 902, the two-way communication devices 1A and 1B are placed on the tables 911 and 912, respectively. However, in FIG. 1B, only the two-way communication device 1A in the conference room 901 is illustrated for simplification of illustration. The same applies to the two-way communication device 1B in the conference room 902. FIG. 2 shows an external perspective view of the two-way communication devices 1A and 1B.
As illustrated in FIG. 1C, a plurality (six in this embodiment) of conference participants A1 to A6 are located around the two-way communication devices 1A and 1B, respectively. However, in FIG. 1C, only conference participants around the two-way communication device 1A in the conference room 901 are illustrated for simplification. The arrangement of conference participants located around the two-way communication device 1B in the other conference room 902 is the same.

The two-way communication device of the present invention can respond by voice via the telephone line 920 between two conference rooms 901 and 902, for example.
Normally, a conversation via the telephone line 920 is performed by one speaker and one speaker, that is, one-on-one, but the two-way communication device of the present invention uses one telephone line 920. A plurality of conference participants A1 to A6 can talk with each other. Although details will be described later, the number of speakers at the same time (same time zone) is limited to one each other in order to avoid voice congestion.
Since the two-way communication device of the present invention is intended for voice (call), only voice is transmitted via the telephone line 920. In other words, a large amount of image data as in the video conference system is not transmitted. Furthermore, since the two-way communication device of the present invention compresses and transmits conference participants' calls, the transmission burden on the telephone line 920 is light.

Configuration of Interactive Communication Device The configuration of an interactive communication device as an embodiment of the present invention will be described with reference to FIGS.
FIG. 2 is a perspective view of a two-way communication device as an embodiment of the present invention.
FIG. 3 is a sectional view of the two-way communication apparatus illustrated in FIG.
4 is a plan view of the microphone / electronic circuit housing portion of the two-way communication apparatus illustrated in FIG. 1, and is a plan view taken along line X-XY in FIG.

As illustrated in FIG. 2, the two-way communication device 1 includes an upper cover 11, a sound reflection plate 12, a connecting member 13, a speaker housing portion 14, and an operation portion 15.
As illustrated in FIG. 3, the speaker housing 14 includes a sound reflecting surface 14 a, a bottom surface 14 b, and an upper sound output opening 14 c. The reception / reproduction speaker 16 is accommodated in a lumen 14d which is a space surrounded by the sound reflection surface 14a and the bottom surface 14b. The sound reflecting plate 12 is positioned above the speaker housing portion 14, and the speaker housing portion 14 and the sound reflecting plate 12 are connected by a connecting member 13.

  A constraining member 17 passes through the connecting member 13, and the constraining member 17 is between the constraining member / lower fixing portion 14 e on the bottom surface 14 b of the speaker housing portion 14 and the constraining member fixing portion 12 b of the sound reflecting plate 12. Is restrained. However, the restraining member 17 only penetrates the restraining member / penetrating portion 14 f of the speaker housing portion 14. The reason why the restraining member 17 penetrates the restraining member / penetrating portion 14f and is not restrained here is that the speaker housing portion 14 vibrates due to the operation of the speaker 16, but the vibration is restrained around the upper sound output opening 14c. This is to prevent it from happening.

The voice spoken by the speaker in the speaker partner conference room is removed from the upper sound output opening 14c through the reception / reproduction speaker 16, and is transmitted between the sound reflecting surface 12a of the sound reflecting plate 12 and the sound reflecting surface 14a of the speaker accommodating portion 14. It spreads in all directions of 360 degrees around the axis CC along the defined space.
As illustrated, the cross section of the sound reflecting surface 12a of the sound reflecting plate 12 depicts a gentle trumpet arc. The cross section of the sound reflecting surface 12a has a cross-sectional shape illustrated over 360 degrees (omnidirectional) about the axis CC.
Similarly, as illustrated in the cross section of the sound reflection surface 14a of the speaker housing portion 14, a gentle convex surface is drawn. The cross section of the sound reflecting surface 14a also has the illustrated cross sectional shape over 360 degrees (omnidirectional) about the axis CC.

The sound S emitted from the reception / reproduction speaker 16 passes through the upper sound output opening 14c, passes through a sound output space having a trumpet-shaped cross section defined by the sound reflection surface 12a and the sound reflection surface 14a, and the voice response device 1 Along the surface of the placed table 911, the sound spreads in all directions 360 degrees around the axis C-C, and is heard at a volume equal to all conference participants A1 to A6. In the present embodiment, the surface of the table 911 is also used as part of the sound propagation means.
The diffusion state of the sound S output from the receiving / reproducing speaker 16 is shown by arrows.

The sound reflecting plate 12 supports the printed circuit board 21.
On the printed circuit board 21, as illustrated in a plan view in FIG. 4, the microphones MC <b> 1 to MC <b> 6, the light emitting diodes LED <b> 1 to 6, the microprocessor 23, the codec (CODEC) 24, and the first digital signal Various electronic circuits such as a processor (DSP 1) DSP 25, a second digital signal processor (DSP 2) DSP 26, an A / D converter block 27, a D / A converter block 28, and an amplifier block 29 are mounted on the sound reflector. Reference numeral 12 also functions as a member that supports the microphone / electronic circuit housing portion 2.

  The printed circuit board 21 has a damper 18 that absorbs vibration from the reception / reproduction speaker 16 so that vibration from the reception / reproduction speaker 16 is transmitted to the sound reflector 12 and does not enter the microphones MC1 to MC6. It is attached. The damper 18 includes a screw and a cushioning material such as an anti-vibration rubber inserted between the screw and the printed board 21, and the cushioning material is screwed to the printed board 21 with a screw. That is, the vibration transmitted from the reception / reproduction speaker 16 to the printed circuit board 21 is absorbed by the buffer material. Thereby, the microphones MC1 to MC6 are not affected by the sound from the speaker 16.

As shown in FIG. 4, six microphones MC <b> 1 to MC <b> 6 are located radially from the central axis C of the printed circuit board 21 at equal intervals (60 degrees in this embodiment). Each microphone is a unidirectional microphone. Its characteristics will be described later.
Each of the microphones MC1 to MC6 is swingably supported by a first microphone support member 22a and a second microphone support member 22b, both of which are flexible or elastic (in order to simplify the illustration, the microphones Only the first microphone support member 22a and the second microphone support member 22b in the MC1 portion are illustrated), and is not affected by the vibration from the reception / reproduction speaker 16 by the damper 18 using the above-described cushioning material. In addition to the countermeasures, the first microphone support member 22a and the second microphone support member 22b having flexibility or elasticity absorb the vibration of the printed circuit board 21 that is vibrated by the vibration from the reception / reproduction speaker 16, and reproduce the reception. The noise of the receiving / reproducing speaker 16 is avoided so as not to be affected by the vibration of the speaker 16.

As illustrated in FIG. 3, the reception / reproduction speaker 16 is oriented perpendicularly to the central axis CC of the plane on which the microphones MC1 to MC6 are located (in the present embodiment, it is directed upward) With the arrangement of the reception / reproduction speaker 16 and the six microphones MC1 to MC6, the distance between the reception / reproduction speaker 16 and each of the microphones MC1 to MC6 is equal. The sound reaches the microphones MC1 to MC6 with almost the same volume and phase. However, due to the configuration of the sound reflecting surface 12a of the sound reflecting plate 12 and the sound reflecting surface 14a of the speaker housing portion 14, the sound of the receiving and reproducing speaker 16 is not directly input to the microphones MC1 to MC6. In addition, as described above, by using the damper 18 using the buffer material, the first microphone support member 22a and the second microphone support member 22b having flexibility or elasticity, the reception / reproduction speaker 16 is provided. The influence of vibration is reduced.
As shown in FIG. 1C, the conference participants A1 to A6 are usually almost equal to the vicinity of the microphones MC1 to MC6 arranged at intervals of 60 degrees in the direction of 360 degrees around the voice response device 1. Located at intervals.

Light- emitting diodes Light-emitting diodes LED1 to 6 are arranged in the vicinity of the microphones MC1 to MC6 as means for reporting that a speaker to be described later has been determined.
The light emitting diodes LED1 to 6 are provided so as to be visible from all the conference participants A1 to A6 even when the upper cover 11 is attached. Therefore, the upper cover 11 is provided with a transparent window so that the light emitting states of the light emitting diodes LED1 to LED6 can be visually recognized. Of course, the upper cover 11 may be provided with openings in the portions of the light emitting diodes LEDs 1 to 6, but a light transmitting window is preferable from the viewpoint of dust prevention to the microphone / electronic circuit housing portion 2.

The printed circuit board 21 includes a first digital signal processor (DSP1) 25, a second digital signal processor (DSP2) 26, and various electronic circuits 27 to 29 for performing various signal processing described later. It is arranged in a space other than the part where MC6 is located.
In the present embodiment, the DSP 25 is used as signal processing means for performing processing such as filter processing and microphone selection processing together with various electronic circuits 27 to 29, and the DSP 26 is used as an echo canceller.

FIG. 5 is a schematic configuration diagram of the microprocessor 23, the codec 24, the DSP 25, the DSP 26, the A / D converter block 27, the D / A converter block 28, the amplifier block 29, and other various electronic circuits.
The microprocessor 23 performs overall control processing of the microphone / electronic circuit housing unit 2. The codec 24 compresses and encodes the voice to be transmitted to the other party conference room.
The DSP 25 performs various signal processing described below, such as filter processing and microphone selection processing.
The DSP 26 functions as an echo canceller.
In FIG. 5, four A / D converters 271 to 274 are illustrated as an example of the A / D converter block 27, and two D / A converters are illustrated as an example of the D / A converter block 28. The converters 281 to 282 are illustrated, and two amplifiers 291 to 292 are illustrated as an example of the amplifier block 29.
In addition, as the microphone / electronic circuit housing portion 2, various circuits such as a power supply circuit are mounted on the printed circuit board 21.

In FIG. 4, a pair of microphones MC1-MC4: MC2-MC5: MC3-M6 arranged in a straight line at symmetrical (or opposite) positions with respect to the central axis C of the printed circuit board 21 each have two channels. The analog signals are input to A / D converters 271 to 273 that convert digital signals. In this embodiment, one A / D converter converts a 2-channel analog input signal into a digital signal. Therefore, the detection signals of two (one pair) microphones, for example, microphones MC1 and MC4, which are positioned on a straight line across the central axis C, are input to one A / D converter and converted into digital signals. Yes. Further, in this embodiment, in order to identify the speaker of the voice to be sent to the other party's conference room, the difference between the two microphones positioned on a straight line, the volume of the voice, etc. are referred to. When the signals of two microphones located on the line are input to the same A / D converter, the conversion timing is also substantially the same, and there is little timing error when taking the difference between the audio outputs of the two microphones. There are advantages such as being easy.
The A / D converters 271 to 274 can also be configured as A / D converters 271 to 274 with a variable gain amplification function.
The collected sound signals of the microphones MC1 to MC6 converted by the A / D converters 271 to 274 are input to the DSP 25, and various signal processing described later is performed.
As one of the processing results of the DSP 25, the result of selecting one of the microphones MC1 to MC6 is output to the light emitting diodes LED1 to LED6.

The processing result of the DSP 25 is output to the DSP 26 and an echo cancellation process is performed. The DSP 26 includes, for example, an echo cancellation transmission processing unit and an echo cancellation reception unit.
The processing result of the DSP 26 is converted into an analog signal by the D / A converters 281 to 282. The output from the D / A converter 281 is encoded by the codec 24 as necessary, and output to the line-out of the telephone line 920 (FIG. 1 (A)) via the amplifier 291 to the partner conference room. The sound is output as a sound through the reception / reproduction speaker 16 of the installed voice response device 1.
Voice from the two-way communication device 1 installed in the other party's conference room is input via the line-in of the telephone line 920 (FIG. 1A), converted into a digital signal by the A / D converter 274, The signal is input to the DSP 26 and used for echo cancellation processing. In addition, the voice from the two-way communication device 1 installed in the other party's conference room is applied to the speaker 16 through a route (not shown) and output as sound.
The output from the D / A converter 282 is output as a sound from the reception / reproduction speaker 16 of the bidirectional communication apparatus 1 via the amplifier 292. In other words, in addition to the voice of the speaker selected in the other party's conference room from the reception / reproduction speaker 16 described above, the conference participants A1 to A6 also use the reception / reproduction speaker 16 for the voice uttered by the speaker in the conference room. Can be heard through.

Microphones MC1 to MC6
FIG. 6 is a graph showing the characteristics of the microphones MC1 to MC6.
Each unidirectional characteristic microphone changes its frequency characteristic and level characteristic as illustrated in FIG. 6 depending on the arrival angle of sound from the speaker to the microphone. The plurality of curves indicate directivity when the frequency of the sound collection signal is 100 Hz, 150 Hz, 200 Hz, 300 Hz, 400 Hz, 500 Hz, 700 Hz, 1000 Hz, 1500 Hz, 2000 Hz, 3000 Hz, 4000 Hz, 5000 Hz, and 7000 Hz. However, in order to simplify the illustration, FIG. 6 typically illustrates the directivity for 150 Hz, 500 Hz, 1500 Hz, 3000 Hz, and 7000 Hz.

7A to 7D are graphs showing the analysis results of the position of the sound source and the sound collection level of the microphone. A speaker is placed at a predetermined distance, for example, a distance of 1.5 meters, from the two-way communication device 1. The result of performing fast Fourier transform (FFT) on the sound collected by each microphone at regular time intervals is shown. The X axis represents frequency, the Y axis represents signal level, and the Z axis represents time.
When the microphone having directivity shown in FIG. 6 is used, strong directivity is shown in front of the microphone. In the present embodiment, using such characteristics, the DSP 25 performs a microphone selection process.

When a non-directional microphone is used instead of a directional microphone as in the present invention, since all sounds around the microphone are collected, the S / N between the voice of the speaker and the ambient noise is confused. Good sound cannot be collected. In order to avoid this, in the present invention, S / N with surrounding noise is improved by collecting sound with one directional microphone.
Furthermore, a microphone array using a plurality of omnidirectional microphones can be used as a method for obtaining the directivity of the microphone. However, in such a method, the time axis (phase) of a plurality of signals is complicated, and thus complicated. Since processing is required, it takes time and response is low, and the apparatus configuration is complicated. That is, the DSP signal processing system also requires complicated signal processing. The present invention solves such a problem by using the directional microphone illustrated in FIG.
Further, in order to synthesize a microphone array signal and use it as a directional sound pickup microphone, there is a disadvantage that the outer shape is restricted by the pass frequency characteristic and the outer shape becomes large. The present invention also solves this problem.

Effects of the device configuration of the communication device The communication device configured as described above exhibits the following advantages.
(1) The positional relationship between the even number of microphones MC1 to MC6 arranged radially at equal angles and at equal intervals and the reception / reproduction speaker 16 is constant, and the distance from the reception / reproduction speaker 16 is very short. The level at which the output sound returns directly to the microphones MC1 to MC6 via the conference room (room) environment is overwhelmingly dominant. Therefore, the characteristics (signal level (intensity), frequency characteristics (f characteristic), phase) that the sound reaches from the speaker 16 to the microphones MC1 to MC6 are always the same. That is, there is an advantage that the two-way communication device 1 in the embodiment of the present invention always has the same transfer function.
(2) Therefore, there is no change in the transfer function when the output of the microphone sent to the other party's conference room is switched when the speakers are different, and there is no need to adjust the gain of the microphone system every time the microphone is switched. Have advantages. In other words, there is an advantage that once the adjustment is made at the time of manufacturing the interactive communication apparatus, it is not necessary to redo the adjustment.
(3) Even if the microphones are switched when the speakers are different for the same reason as described above, only one echo canceller (DSP 26) is required. The DSP is expensive, and it is not necessary to arrange a plurality of DSPs on the printed circuit board 21 on which various members are mounted and the space is small, and the space for arranging the DSPs on the printed circuit board 21 may be small. As a result, the printed circuit board 21, and thus the communication device of the present invention can be made smaller.
(4) As described above, since the transfer function between the reception and reproduction speaker 16 and the microphones MC1 to MC6 is constant, for example, the sensitivity difference adjustment of the microphone itself having ± 3 dB can be performed by the microphone unit of the two-way communication device alone. There is an advantage. Details of the sensitivity difference adjustment will be described later.
(5) The table on which the two-way communication device 1 is mounted normally uses a round table or a polygonal table. However, a single reception / reproduction speaker 16 in the two-way communication device 11 can provide sound of equal quality. A loudspeaker system that is uniformly distributed (diffused) in all directions of 360 degrees around the axis C has become possible.
(6) The sound emitted from the receiving / reproducing speaker 16 is transmitted to the table surface of the round table (boundary effect), effectively and evenly delivering high-quality sound to the conference participants, and facing the ceiling direction of the conference room There is an advantage that the sound and phase on the side are canceled and become a small sound, and there are few reflected sounds from the ceiling direction to the conference participants, and as a result, a clear sound is distributed to the participants.
(7) Since the sound emitted from the reception / reproduction speaker 16 reaches all the microphones MC1 to MC6 arranged radially and at equal intervals at the same angle at the same volume at the same time, it is determined whether the sound is the voice of the speaker or the received voice. It becomes easy. As a result, erroneous determination of microphone selection processing is reduced. Details thereof will be described later.
(8) Even number, for example, six microphones are arranged at equal angles radially and at equal intervals, and a pair of opposing microphones are arranged in a straight line, so that level comparison for direction detection can be easily performed.
(9) By the damper 18, the microphone support member 22, and the like, it is possible to reduce the influence of the vibration due to the sound of the reception and reproduction speaker 16 on the sound collection of the microphones MC1 to MC6.
(10) As illustrated in FIG. 3, structurally, the sound of the reception / reproduction speaker 16 does not directly propagate to the microphones MC1 to MC6. Therefore, in the two-way communication apparatus 1, the influence of noise from the reception / reproduction speaker 16 is small.

The communication device 1 described with reference to FIGS. 2 to 3 has the reception reproduction speaker 16 disposed in the lower portion and the microphones MC1 to MC6 (and related electronic circuits) disposed in the upper portion. The positions of 16 and microphones MC1-MC6 (and associated electronic circuitry) can also be turned upside down as illustrated in FIG. Even in such a case, the above-described effects are exhibited.

  The number of microphones is not limited to six, and any number of microphones such as four, eight, etc. may be arranged in a straight line (in the same direction) with a plurality of pairs of axes C radially and equally spaced at the same angle. They are arranged in a straight line like MC1 and MC4. The reason why the two microphones MC1 and MC4 are arranged to face each other is to select a microphone and specify a speaker.

Signal Processing Contents Hereinafter, processing contents mainly performed by the first digital signal processor (DSP) 25 will be described.
FIG. 9 is a diagram illustrating an outline of processing performed by the DSP 25. The outline is described below.

(1) Measurement of ambient noise As an initial operation, preferably, ambient noise where the two-way communication device 1 is installed is measured.
The two-way communication device 1 can be used in various environments (conference rooms). In order to improve the performance of the two-way communication device 1 in order to ensure the accuracy of selection of the microphone, in the present invention, noise in the surrounding environment where the two-way communication device 1 is installed is measured in the initial stage. It is possible to eliminate the influence from the signal collected by the microphone.
Of course, when the two-way communication apparatus 1 is repeatedly used in the same conference room, noise measurement is performed in advance, and this process can be omitted when the noise state does not change.
Note that noise measurement can also be performed in a normal state.
Details of the noise measurement will be described later.

(2) Selection of Chairperson For example, when the two-way communication device 1 is used for a two-way conference, it is beneficial to have a chairman who manages the proceedings in each conference room. Therefore, as one aspect of the present invention, the chairperson is set from the operation unit 15 of the interactive communication device 1 in the initial stage of using the interactive communication device 1. As a chairperson setting method, for example, the first microphone MC1 located in the vicinity of the operation unit 15 is used as a chairperson microphone. Of course, the chairman's microphone can be arbitrary.
Note that this processing can be omitted when the chairperson who repeatedly uses the interactive communication device 1 is the same. Or you may decide the microphone of the position where a chairperson sits beforehand. In that case, there is no need to select a chairman each time.
Of course, the selection of the chair is not limited to the initial state, and can be performed at any timing.
Details of the chairperson selection will be described later.

(3) Microphone sensitivity difference adjustment As an initial operation, preferably, the gain or attenuation unit of the amplification unit that amplifies the signals of the microphones MC1 to MC6 so that the acoustic coupling between the reception reproduction speaker 16 and the microphones MC1 to MC6 is equal. Automatically adjust the attenuation value.
The sensitivity difference adjustment will be described later.

Various processes exemplified below are performed as normal processes.
(4) Microphone selection / switching process When a plurality of conference participants make a call at the same time in one conference room, voices are mixed and difficult for the conference participants A1 to A6 in the other conference room. Therefore, in the present invention, in principle, one person is allowed to talk at a time. For this reason, the DSP 25 performs microphone selection / switching processing.
As a result, only the call from the selected microphone is transmitted to the voice response device 1 in the other party conference room via the telephone line 920 and output from the speaker. Of course, as described with reference to FIG. 5, the LED in the vicinity of the selected speaker's microphone is turned on, and the selected speaker's voice is also output from the speaker of the interactive communication device 1 in the room. Can hear and recognize who is an authorized speaker.
The purpose of this processing is to select a signal from a unidirectional microphone facing the speaker and send a signal having a good S / N to the other party as a transmission signal.
(5) Display of the selected microphone Lights so that all the conference participants A1 to A6 can easily recognize which conference participant's microphone is selected and allowed to speak. The corresponding ones of the diodes LEDs 1 to 6 are turned on.
(6) As a background art of the microphone selection process described above, or in order to accurately perform the microphone selection process, various signal processes exemplified below are performed.
(A) Band separation and level conversion processing of microphone collected signal (b) Start / end determination processing of speech
To be used as a trigger to start selecting the microphone signal that faces the speaker direction.
(C) Speaker direction microphone detection processing
To analyze the collected sound signal of each microphone and determine the microphone used by the speaker.
(D) Speaker direction microphone switching timing determination processing, and microphone signal selection switching processing facing the detected speaker
An instruction to switch to the microphone selected from the above processing result is given. (E) Measurement of floor noise during normal operation

Measurement of floor (environment) noise This process is divided into an initial process and a normal process immediately after the two-way communication device is turned on.
This process is performed under the following exemplary preconditions.

Immediately after turning on the power of the interactive communication apparatus 1, the DSP 25 performs the following noise measurement described with reference to FIGS.
The initial processing of the DSP 25 immediately after turning on the power of the two-way communication device 1 is to measure the floor noise and the reference signal level, and based on the difference between them, a guideline of the effective distance between the speaker and the present system and the speech start / end determination threshold level. To set up.
The level value peak-held by the sound pressure level detection unit in the DSP 25 is read at a constant time interval, for example, 10 mSec, and an average value of unit time values is calculated and used as floor noise. Then, the DSP 25 determines a threshold value for a speech start detection level and a speech end detection level based on the measured floor noise level.

FIG. 10, Process 1: Test Level Measurement The DSP 25 outputs a test tone to the line-in terminal of the reception signal system illustrated in FIG. 5 according to the process illustrated in FIG. The sound is collected at ~ MC6, and the average value is obtained using the signal as a speech start reference level.

FIG. 11, Process 2: Noise measurement 1
In accordance with the process illustrated in FIG. 11, the DSP 25 collects the level of the collected sound signal from each of the microphones MC1 to MC6 as a floor noise level for a predetermined time and obtains an average value.

FIG. 12, Process 3: Effective distance trial DSP 25 compares the speech start reference level with the floor noise level according to the process illustrated in FIG. And the effective distance between the speaker who works well in the two-way communication device 1 and the two-way communication device 1 is calculated.

As a result of the microphone selection prohibition determination process 3, if the floor noise is larger (higher) than the speech start reference level, the DSP 25 determines that there is a strong noise source in the direction of the microphone, and automatically selects a microphone in that direction. The prohibition is set, and this is displayed, for example, on the light emitting diodes LED1 to 6 or the operation unit 15.

As illustrated in FIG. 13, the threshold value determination DSP 25 compares the speech start reference level and the floor noise level, and determines the threshold values of the speech start and end levels from the difference.

  As far as noise measurement is concerned, the next process is a normal process, so the DSP 25 sets each timer (counter) and prepares for the next process.

The noise normal processing DSP 25 performs noise processing according to the processing shown in FIG. 14 in the normal operation state after the noise measurement during the initial operation of the two-way communication device 1, and each of the six microphones MC1 to MC6. The volume level average value of the selected speaker and the noise level after detection of the end of the speech are measured, and the speech start / end determination threshold level is reset in a fixed time unit.

FIG. 14, Process 1 : The DSP 25 determines branching to Process 2 or Process 3 based on the determination of whether the speech is in progress or the end of speech.

FIG. 14, Process 2 : Speaker Level Measurement The DSP 25 averages and records the level data for a unit time, for example, 10 seconds, for a plurality of times, for example, 10 times, as a speaker level.
If the utterance ends within the unit time, the time measurement and the utterance level measurement are stopped until a new utterance starts, and the measurement process is resumed after the new utterance is detected.

FIG. 14, Process 3 : Floor noise measurement 2
The DSP 25 averages the noise level data for a unit time, for example, 10 seconds from the detection of the end of the speech to the start of the speech, and records the average as a floor noise level a plurality of times, for example, 10 times.
If there is a new message within the unit time, the DSP 25 stops the time measurement and noise measurement on the way, and restarts the measurement process after detecting the end of the new message.

FIG. 14, Process 4 : Threshold Determination 2
The DSP 25 compares the speech level and the floor noise level, and determines the threshold values for the speech start and end levels from the difference.
In addition to this, since the average value of the speaking level of the speaker is obtained, the speaking start and end detection threshold levels specific to the speaking party facing the microphone can be set.

Generation of Various Frequency Component Signals by Filter Processing FIG. 15 is a configuration diagram showing filtering processing performed by the DSP 25 using sound signals collected by a microphone as preprocessing. FIG. 15 shows processing for one microphone (channel (one sound collection signal)).
The collected sound signal of each microphone is processed by an analog low cut filter 101 having a cutoff frequency of 100 Hz, for example, and a filtered audio signal from which a frequency of 100 Hz or less has been removed is output to the A / D converter 102. , Digital high-cut filters 103a to 103e (collectively referred to as “collection signals”) having cut-off frequencies of 7.5 KHz, 4 KHz, 1.5 KHz, 600 Hz, and 250 Hz, respectively. 103), high frequency components are removed (high cut processing). The results of the digital high cut filters 103a to 103e are further subtracted for each filter signal of the adjacent digital high cut filters 103a to 103e in subtractors 104a to 104d (collectively 104).
In the embodiment of the present invention, the digital high cut filters 103a to 103e and the subtractors 104a to 104d are actually processed in the DSP 25. The A / D converter 102 can be realized as one of the A / D converter blocks 27.

  FIG. 16 is a frequency characteristic diagram showing the filter processing result described with reference to FIG. Thus, a plurality of signals having various frequency components are generated from a signal collected by a microphone having one directivity.

The start and end of speech is determined as one of the triggers for starting the band-pass filter processing and microphone signal level conversion processing microphone selection processing. A signal used for this purpose is obtained by the bandpass filter processing and level conversion processing illustrated in FIG. FIG. 17 shows only 1CH during input signal processing of 6 channels (CH) collected by the microphones MC1 to MC6.
The band-pass filter processing and level conversion processing unit in the DSP 25 respectively collects the collected sound signals of the microphones of each channel at 100 to 600 Hz, 200 to 250 Hz, 250 to 600 Hz, 600 to 1500 Hz, 1500 to 4000 Hz, 4000 to 7500 Hz. Band-pass filters 201a to 201a having band-pass characteristics (collectively, band-pass filter block 201), original microphone sound collection signals, and level converters 202a to 202g (for level conversion of the band-pass sound collection signals) Collectively, it has a level conversion block 202).

  Each level conversion unit 202 a to 202 g includes a signal absolute value processing unit 203 and a peak hold processing unit 204. Therefore, as illustrated in the waveform diagram, the signal absolute value processing unit 203 inverts the sign and converts it to a positive signal when a negative signal indicated by a broken line is input. The peak hold processing unit 204 holds the maximum value of the output signal of the signal absolute value processing unit 203. However, in the present embodiment, the held maximum value is somewhat lowered with the passage of time. Of course, the peak hold processing unit 204 can be improved so that the maximum value can be held for a long time by reducing the decrease.

A bandpass filter will be described. The band-pass filter used for the two-way communication device 1 is composed of, for example, a secondary IIR high-cut filter and a microphone signal input stage low-cut filter only.
In the present embodiment, it is utilized that if the signal that has passed through the high-cut filter is subtracted from the signal having a flat frequency characteristic, the rest is substantially equivalent to the signal that has passed through the low-cut filter.
In order to match the frequency-level characteristics, an extra band-pass bandpass filter is required for one band, but the band required by the number of filter stages equal to the number of bands of the required bandpass filter + 1 and the filter coefficient A pass is obtained. The band frequency of the hand pass filter required this time is the following 6 band pass filter per channel (CH) of the microphone signal.

In this method, the calculation program of the above IIR filter in the DSP 25 is only 6CH (channel) × 5 (IIR filter) = 30.
Contrast with the conventional band-pass filter configuration. Assuming that the band-pass filter uses a second-order IIR filter and a 6-band band-pass filter is prepared for each of six microphone signals as in the present invention, in the conventional method, 6 × 6 × 2 = 72. Circuit IIR / filtering is required. This processing requires considerable program processing even with the latest excellent DSP, and affects other processing.
In the embodiment of the present invention, the 100 Hz low cut filter is processed by an analog filter in the input stage. There are five types of cutoff frequencies of the prepared second-order IIR high cut filters: 250 Hz, 600 Hz, 1.5 KHz, 4 KHz, and 7.5 KHz. Of these, the high-cut filter with a cutoff frequency of 7.5 kHz is not necessary because the sampling frequency is actually 16 KHz. Deliberately rotate the phase of the attenuator to reduce the phenomenon.

  FIG. 18 is a flowchart when processing by the DSP 25 is performed according to the configuration illustrated in FIG.

  In the DSP 25 illustrated in FIG. 18, a high-pass filter process is performed as the first stage process, and a subtraction process from the result of the first-stage high-pass filter process is performed as the second stage process. FIG. 16 is an image frequency characteristic diagram of the signal processing result. [X] below shows each processing case in FIG.

First stage [1] The input signal is passed through a 7.5 kHz high cut filter for the whole band pass filter. This filter output signal becomes a bandpass filter output of [100Hz-7.5KHz] by matching the analog low cut of the input.

  [2] Pass the input signal through a 4KHz high cut filter. This filter output signal becomes a bandpass filter output of [100Hz-4KHz] by combining with the input analog low cut filter.

  [3] Pass the input signal through a 1.5 kHz high cut filter. This filter output signal is combined with the input analog low cut filter [100Hz-1.5KHz] is combined with the input analog low cut filter [100Hz-1.5KHz] When combined with the input analog low cut filter [100Hz -1.5KHz] bandpass filter output.

  [4] Pass the input signal through a 600 kHz high cut filter. This filter output signal becomes a bandpass filter output of [100Hz-600Hz] by combining with the input analog low cut filter.

  [5] The input signal is passed through a 250 kHz high cut filter. This filter output signal becomes a bandpass filter output of [100Hz-250Hz] by combining with the input analog low cut filter.

The second stage [1] band pass filter (BPF5 = [4KHz ~ 7.5KHz]) executes the process of filter output [1]-[2] ([100Hz ~ 7.5KHz]-[100Hz ~ 4KHz]) The signal output is [4KHz to 7.5KHz].
[2] The bandpass filter (BPF4 = [1.5KHz to 4KHz]) will perform the above processing when the filter output [2]-[3] ([100Hz to 4KHz]-[100Hz to 1.5KHz]) is executed. Output [1.5KHz ~ 4KHz].
[3] The bandpass filter (BPF3 = [600Hz to 1.5KHz]) performs the above processing when the filter output [3]-[4] ([100Hz to 1.5KHz]-[100Hz to 600Hz]) is executed. Output [600Hz ~ 1.5KHz].
[4] The bandpass filter (BPF2 = [250Hz to 600Hz]) performs the process of filter output [4]-[5] ([100Hz to 600Hz]-[100Hz to 250Hz]). ~ 600Hz]. [5] The bandpass filter (BPF1 = [100 Hz to 250 Hz]) uses the signal [5] as it is as the output signal [5].
[6] The bandpass filter (BPF6 = [100 Hz to 600 Hz]) uses the signal [4] as it is and outputs it as the output signal (4).
The bandpass filter output required by the above processing in the DSP 25 is obtained.

  The input microphone sound collection signals MIC1 to MIC6 are constantly updated in the DSP 25 as the sound pressure level of the entire band and the sound pressure level of the six bands that have passed through the bandpass filter as shown in Table 5.

In Table 5, for example, L1-1 indicates a peak level when the collected sound signal of the microphone MC1 passes through the first bandpass filter 201a.
The start and end of speech is determined using a microphone sound collection signal that has passed through the 100 Hz to 600 Hz bandpass filter 201a shown in FIG. 17 and whose sound pressure level has been converted by the level converter 202b.

  The conventional band-pass filter is configured by combining a high-pass filter and a low-pass filter for each stage of the band-pass filter. Therefore, a 36-band band-pass filter of the specification used in this embodiment is used. When constructed, 72 circuits of filter processing are required. In contrast, the filter configuration of the embodiment of the present invention is simplified as described above.

Sentence start / end determination processing The first digital signal processor (DSP1) 25, based on the value output from the sound pressure level detector, as shown in FIG. If it rises and exceeds the threshold of the speech start level, it is determined that the speech starts.If the level continues to be higher than the threshold of the start level, the floor noise is determined if the level is lower than the threshold of speech end during speech. The speech end determination time is determined, for example, when it is continued for 0.5 seconds, the speech end is determined.
The start and end of speech is determined based on sound pressure level data (microphone signal level (1)) that has passed through a 100 Hz to 600 Hz bandpass filter whose sound pressure level has been converted by the microphone signal conversion processing unit 202b illustrated in FIG. It is determined that the utterance has started when the threshold level illustrated in FIG. 19 is reached.
In order to avoid malfunction due to frequent microphone switching, the DSP 25 does not detect the next speech start after the speech start determination time, for example, 0.5 seconds.

The microphone selection DSP 25 performs speaker direction detection and automatic selection of a microphone signal facing the speaker in the mutual communication system based on a so-called “star chart method”.
FIG. 20 is a graph illustrating the operation mode of the interactive communication device 1.
FIG. 21 is a flowchart showing normal processing of the interactive communication device 1.

As illustrated in FIG. 20, the two-way communication device 1 performs voice signal monitoring processing according to the collected sound signals from the microphones MC1 to MC6, performs speech start / end determination, performs speech direction determination, and selects a microphone. The result is displayed on the light emitting diodes LED1 to LED6.
The operation will be described below with the DSP 25 in the two-way communication device 1 as a main component with reference to the flowchart of FIG. The overall control of the microphone / electronic circuit housing unit 2 is performed by the microprocessor 23, and the processing of the DSP 25 will be mainly described.

Step 1: Level Conversion Signal Monitoring Signals collected by the microphones MC1 to MC6 are respectively obtained in the band-pass filter block 201 and the level conversion block 202 described with reference to FIGS. Therefore, the DSP 25 constantly monitors seven types of signals for each microphone sound collection signal.
Based on the monitoring result, the DSP 25 proceeds to any one of the speaker direction detection processing 1, the speaker direction detection processing 2, and the speech start / end determination processing.

Step 2: Speech Start / End Determination Processing The DSP 25 determines the start and end of speech according to the method described in detail below with reference to FIG. When the DSP 25 detects the start of speech, the DSP 25 informs the speaker direction determination processing in step 4 of the start of speech.
When the speech start / end determination process in step 2 is performed, when the speech level becomes lower than the speech end level, a speech end determination time (for example, 0.5 second) timer is activated and the speech end determination time and the speech level are When the level is lower than the end level, it is determined that the speech has ended.
If it becomes larger than the speech end level within the speech end determination time, it waits until it becomes smaller than the speech end level again.

Step 3: Speaker Direction Detection Processing The speaker direction detection processing in the DSP 25 is continuously performed by continuously searching for the speaker direction. Thereafter, the data is supplied to the speaker direction determination processing in step 4.

Step 4: Speaker direction microphone switching processing The timing determination processing in the speaker direction microphone switching processing in the DSP 25 has been selected from the results of the processing in step 2 and step 3 and the speaker detection direction at that time. If the speaker direction is different, the microphone selection in step 4 is instructed to select a microphone in a new speaker direction.
However, if the chairman's microphone is set from the operation unit 15 and the chairman's microphone and another conference participant speak at the same time, the chairman's comment is given priority.
At this time, the selected microphone information is displayed on the light emitting diodes LED1 to LED6.

Step 5: Transmission of microphone sound collecting signal In the microphone signal switching process, only the microphone signal selected by the process of Step 4 from the six microphone signals is used as the transmission signal, and the two-way communication device 1 through the telephone line 920. For transmission to the other party's two-way communication device, the data is output to the line-out of the telephone line 920 illustrated in FIG.

Processing for setting a speech start level threshold and a speech end threshold 1: Immediately after turning on the power, the floor noise for a predetermined time, for example, 1 second, of each microphone is measured.
The DSP 25 reads out the peak-held level value of the sound pressure level detection unit at a constant time interval, for example, 10 mSec interval in this embodiment, and calculates an average value of values for a predetermined time, for example, 1 minute to obtain floor noise. And
The DSP 25 determines a speech start detection level (floor noise +9 dB) and a speech end detection level threshold (floor noise +6 dB) based on the measured floor noise level. The DSP 25 thereafter reads the peak-held level value of the sound pressure level detector at regular time intervals.
When it is determined that the speech has ended, the DSP 25 functions as a floor noise measurement, detects the start of speech, and updates the threshold for the detection level of speech end.

  According to this method, since the floor noise level at the position where the microphone is placed is different in this threshold setting, a threshold can be set for each microphone, and erroneous determination in selection of the microphone by the noise source can be prevented.

Process 2: Dealing with ambient noise (large floor noise) rooms Process 2 performs the following as a countermeasure when it is difficult to detect the start and end of speech when the floor level is high in Process 1 and the threshold level is automatically updated. .
The DSP 25 determines a threshold for the detection level of the speech start and the detection level of the speech end based on the predicted floor noise level.
The DSP 25 sets the speech start threshold level to be greater than the speech end threshold level (for example, a difference of 3 dB or more).
The DSP 25 reads the level value peak-held by the sound pressure level detector at regular time intervals.

  According to this method, since the threshold value is the same value for all microphones, the person who is behind the noise source and the person who is not so have the same voice volume and can recognize the start of speech.

Talk start judgment
Process 1 : The output level of the sound pressure level detector corresponding to the six microphones is compared with the threshold value of the speech start level.
When the output level of the sound pressure level detector corresponding to all the microphones exceeds the threshold of the speech start level, the DSP 25 determines that the signal is from the reception / reproduction speaker 16 and does not determine that the speech is started. This is because the distance between the reception / reproduction speaker 16 and all the microphones MC1 to MC6 is the same, so that the sound from the reception / reproduction speaker 16 reaches almost all the microphones MC1 to MC6.

Process 2 : Two unidirectional microphones (with microphones MC1 and MC1) with the directional axes shifted by 180 degrees in the opposite direction at an equal angle of 60 degrees with respect to the six microphones illustrated in FIG. Three sets of MC4, microphones MC2 and MC5, and microphones MC3 and MC6) are used, and the level difference of the microphone signal is used. That is, the following calculation is performed.

The DSP 25 compares the absolute values [1], [2], and [3] with the threshold value of the speech start level, and determines that the speech is started when the threshold value of the speech start level is exceeded.
In the case of this process, since all the absolute values do not become larger than the threshold value of the speech start level as in process 1 (because the sound from the reception / reproduction speaker 16 reaches all the microphones equally), the reception / reproduction speaker 16 It is not necessary to determine whether the sound is from the speaker or from the speaker.

Speaker Direction Detection Processing For detecting the speaker direction, the characteristics of the unidirectional microphone illustrated in FIG. 6 are used. As illustrated in FIG. 6, the frequency characteristics and level characteristics of the unidirectional microphone change depending on the sound arrival angle from the speaker to the microphone. The results are illustrated in FIGS. 7 (A) to (D). FIGS. 7A to 7D show a fast Fourier transform (FFT) at a predetermined time interval for the sound collected by each microphone with a speaker placed at a predetermined distance from the two-way communication device 1, for example, a distance of 1.5 meters. ) Result. The X axis represents frequency, the Y axis represents signal level, and the Z axis represents time. The horizontal line represents the cut-off frequency of the band-pass filter, and the level of the frequency band sandwiched between the lines is the 5-band band pass from the microphone signal level conversion processing described with reference to FIGS.・ Data converted to sound pressure level through the filter.

A determination method applied as an actual process for detecting the direction of the speaker in the two-way communication device 1 as one embodiment of the present invention will be described.
Appropriate weighting processing is performed on the output level of each band-pass filter (0 for 1 dB full span (1 dBFs) step, 0 for 0 dBFs, 3 for -3 dBFs, or vice versa). This weighting step determines the processing resolution.
The above weighting process is executed for each sample clock, and the weighted score of each microphone is added and averaged with a fixed number of samples, and the microphone signal having a small (large) total score is determined as a microphone facing the speaker. To do. Table 7 below is an image of this result.

In this example illustrated in Table 7, the smallest total point is the first microphone MC1, so the DSP 25 determines that there is a sound source in the direction of the first microphone MC1 (there is a speaker). The DSP 25 holds the result in the form of a sound source direction microphone number.
As described above, the DSP 25 performs weighting on the output level of the bandpass filter in the frequency band for each microphone, and ranks the microphone signals in the order of smaller (or larger) scores of the output of each bandbandpass filter. The microphone signal having the first rank in three or more bands is determined as the microphone facing the speaker. Then, the DSP 25 creates a score table as shown in Table 8 below assuming that there is a sound source in the direction of the first microphone MC1 (there is a speaker).

  Actually, the performance of the first microphone MC1 is not necessarily the best in the output of all bandpass filters due to the reflection of sound and the influence of standing waves depending on the characteristics of the room, but the majority in the 5 bands If it is 1st place, it can be determined that there is a sound source in the direction of the first microphone MC1 (there is a speaker). The DSP 25 holds the result in the form of a sound source direction microphone number.

  The DSP 25 sums the output level data of each band band pass filter of each microphone in the form shown in Table 9 below, determines that the microphone signal having a high level is the microphone facing the speaker, and determines the result as the sound source direction microphone number. Hold in the form of.

Talker direction microphone switching timing determination processing When activated by the speech start determination result of step 2 in FIG. 21 and when a new speaker microphone is detected from the speaker direction detection processing result of step 3 and past selection information, The DSP 25 issues a microphone signal switching command to the microphone signal selection switching process in step 5 and notifies the light emitting diodes LED1 to 6 that the speaker microphone has been switched. Informs that the communication device 1 has responded.

In order to eliminate the influence of reflected sound and standing waves in a room with high reverberation, the DSP 25 prohibits the activation of a new microphone selection command if the speech end determination time (for example, 0.5 seconds) has not elapsed after switching the microphone.
In this embodiment, two microphone selection switching timings are prepared from the result of the microphone signal level conversion process in step 1 in FIG. 21 and the result of the speaker direction detection process in step 3.

First method : When it is possible to clearly determine the start of speech When speech from the direction of the selected microphone has ended and there is a new speech from another direction.
In this case, the DSP 25 starts speaking after all the microphone signal level (1) and the microphone signal level (2) are equal to or lower than the speech end threshold level and more than the speech end determination time (for example, 0.5 seconds). When any microphone signal level (1) is equal to or higher than the speech start threshold level, it is determined that speech has started, and a microphone facing the speaker direction is properly collected based on the information of the microphone number in the sound source direction. The microphone is determined, and the microphone signal selection switching process in step 5 is started.

Second method : When there is a new louder voice from another direction while the voice is continuing In this case, the DSP 25 starts the voice from the start of the voice (when the microphone signal level (1) exceeds the threshold level) and the voice ends. The determination process starts after the determination time (for example, 0.5 seconds) has elapsed. If it is determined that the sound source direction microphone number from the process 3 is changed and is stable before the end of the speech is detected, the DSP 25 is more than the speaker currently selected for the microphone corresponding to the sound source direction microphone number. It is determined that there is a speaker who is speaking loudly, the sound source direction microphone is determined as a valid sound collecting microphone, and the microphone signal selection switching process in step 5 is started.

The microphone signal selection switching processing DSP 25 facing the detected speaker is activated by the command selected and determined by the command from the speaker direction microphone switching timing determination processing in step 4 of FIG.
The microphone signal selection switching process of the DSP 25 is composed of a 6-circuit multiplier and a 6-input adder as illustrated in FIG. In order to select the microphone signal, the DSP 25 sets the channel gain (channel gain: CH Gain) of the multiplier to which the microphone signal to be selected is connected to [1], and the CH gains of the other multipliers to [0]. By doing so, the selected signal of (microphone signal × [1]) and the processing result of (microphone signal × [0]) are added to the adder, and a desired microphone selection signal is obtained at the output.

  When the channel gain is switched between [1] and [0] as described above, there is a possibility that a click sound is generated due to the level difference of the microphone signal to be switched. Therefore, in the two-way communication device 1, as illustrated in FIG. 23, in order to change the change in CH Gain from [1] to [0] and from [0] to [1], for example, By continuously changing and crossing in a time of 10 milliseconds, the generation of a click sound due to the difference in the level of the microphone signal is avoided.

  Further, by setting the maximum channel gain to other than [1], for example, [0.5], the echo cancellation processing operation in the DSP 25 at the subsequent stage can be adjusted.

  As described above, the call device according to the embodiment of the present invention is not affected by noise, and can be effectively applied to a call device such as a conference.

The communication device according to the embodiment of the present invention has the following advantages in terms of structure.
(1) The positional relationship between a plurality of microphones having a single directivity and the reception / reproduction speaker is constant, and furthermore, since the distance is very close, the sound emitted from the reception / reproduction speaker passes through the conference room (room) environment. The level that returns directly to the multiple microphones is overwhelmingly dominant. Therefore, the characteristics (signal level (intensity), frequency characteristics (f characteristic), phase) for sound to reach a plurality of microphones from the receiving / reproducing speaker are always the same. That is, there is an advantage that the transfer function is always the same in the communication device.

  (2) Therefore, there is no change in the transfer function when the microphone is switched, and there is an advantage that it is not necessary to adjust the gain of the microphone system every time the microphone is switched. In other words, there is an advantage that it is not necessary to start over once the adjustment is made at the time of manufacturing the communication device.

  (3) Even if the microphone is switched for the same reason as described above, only one echo canceller configured by a digital signal processor (DSP) may be used. The DSP is expensive, and the space for placing the DSP on a printed circuit board on which various members are mounted and there is little space may be small.

  (4) Since the transfer function between the receiving / reproducing speaker and the plurality of microphones is constant, there is an advantage that the sensitivity difference of the microphone itself having ± 3 dB can be adjusted by the unit alone. (5) The table on which the communication device is mounted normally uses a round table, but a speaker system that evenly distributes sound of equal quality in all directions with one reception / reproduction speaker in the communication device can be realized. became.

  (6) The sound emitted from the receiving / reproducing speaker is transmitted to the table surface (boundary effect), and the sound is effectively and evenly delivered to the conference participants, and the sound on the opposite side to the ceiling of the conference room. The phase is canceled to produce a small sound, and there is an advantage that the conference participant has less reflected sound from the ceiling direction, and as a result, a clear sound is distributed to the participant.

  (7) Since the sound emitted from the reception / reproduction speaker reaches all of the plurality of microphones at the same volume at the same time, it is easy to determine whether the sound is the speaker's voice or the reception voice. As a result, erroneous determination of microphone selection processing is reduced.

  (8) By arranging even number of microphones at equal intervals, level comparison for direction detection can be easily performed.

  (9) Due to a damper using a cushioning material, a microphone support member having flexibility or elasticity, vibration due to the sound of the reception and reproduction speaker that can be transmitted through the printed circuit board on which the microphone is mounted is collected by the microphone. Can reduce the influence.

  (10) The sound of the receiving / reproducing speaker does not directly enter the microphone. Therefore, in this two-way communication device, the influence of noise from the reception / reproduction speaker is small.

The above communication device has the following advantages from the signal processing aspect.
(A) A plurality of unidirectional microphones are arranged radially at equal intervals so that the direction of the sound source can be detected, and the microphone signal is switched to collect (collect) sound with good S / N and clear sound. Can be sent to the other party.
(B) Sound from surrounding speakers can be collected with good S / N and a microphone facing the speaker can be automatically selected.
(C) In the present invention, signal analysis is simplified by dividing a passing voice frequency band as a method of microphone selection processing and comparing levels of the divided frequency bands.
(D) The microphone signal switching processing of the present invention is realized as DSP signal processing, and a plurality of signals are all cross-fade processed so as not to generate a clicking sound at the time of switching.
(E) The microphone selection result can be notified to display means such as a light emitting diode or to the outside. Therefore, for example, it can be used as speaker position information for a television camera.

The configuration and processing operation of the interactive communication device 1 have been described in detail above.
A technique that applies the above-described effects and features of the interactive communication apparatus 1 is an imaging apparatus that is an embodiment of the present invention described later.
In other words, the imaging apparatus according to the present invention is applied to the imaging apparatus in that the characteristic point of the above-described two-way communication apparatus 1, that is, the even number of microphones are arranged at equal intervals and the sound generation direction can be detected. Technology.

Embodiment FIG. 24 is a diagram for explaining a configuration of an imaging apparatus 40 in the present embodiment.
As shown in FIG. 24, the imaging apparatus 40 includes two or more sound source direction detection units 50a and 50b as microphone detection means, imaging units 60a and 60b as imaging means, and a sound source position estimation unit 70 as sound source position specifying means. And an image recognition unit 80 as image recognition means.
The imaging target of the imaging device 40 is a room R1 in a range surrounded by a line in FIG. 24, and sound source direction detection units 50a and 50b and imaging units 60a and 60b are arranged in the imaging target room R1.
The sound source position estimating unit 70 and the image recognizing unit 80 may also be arranged in the room R1 to be imaged, but this is not always necessary, so in FIG. 24 they are arranged outside the room R1.

  The sound source direction detection units 50a and 50b and the sound source position estimation unit 70 are electrically connected to each other, and the imaging units 60a and 60b and the sound source position estimation unit 70 are electrically connected to each other. Furthermore, the imaging units 60a and 60b and the image recognition unit 80 are electrically connected to each other.

It is possible to freely set how the sound source direction detection units 50a and 50b and the imaging units 60a and 60b are arranged in the room R1 that is the imaging target. However, it is desirable that the imaging units 60a and 60b be arranged at opposite positions in the room R1, as shown in FIG. 24, from the viewpoint of enabling imaging of the room R1 to be imaged as much as possible.
Further, the sound source direction detection units 50a and 50b are preferably arranged at positions as far as possible from each other so as to specify the sound source direction over a wide range.

Hereinafter, each component of the imaging device 40 will be described.
Sound source direction detection units 50a and 50b
The sound source direction detectors 50a and 50b are the same as those of the bidirectional communication device 1 already described in detail. That is, six microphones are equally arranged in all directions at intervals of 60 degrees, and a level comparison for detecting the direction of a sound source is performed based on an audio signal input by each microphone, and a microphone with a high level is selected.
The selected microphones are held as sound source direction microphone numbers MC_a and MC_b in the DSPs 25a and 25b mounted on the sound source direction detection units 50a and 50b.
The sound source direction microphone numbers MC_a and MC_b selected by the sound source direction detecting units 50a and 50b are supplied to the sound source position estimating unit 70 described later as signals S57a and S57b shown in FIG.

Imaging units 60a and 60b
The imaging units 60a and 60b are configured to include a camera with a zoom lens as imaging means capable of capturing a moving image. For example, a digital camera including an imaging device such as a CCD is included, and the captured moving images are sequentially transferred to the image recognition unit 80 as imaging signals S68a and S68b, respectively.
The imaging units 60a and 60b are configured to be capable of rotating in the horizontal direction in FIG. 24 so that the entire area of the room R1 that is the imaging target can be imaged.
And the imaging parts 60a and 60b are provided with rotation means (not shown), such as a stepping motor, and a motor control circuit, for example.

The motor control circuit controls the motor to determine the horizontal position of the imaging units 60a and 60b based on S76a and S76b respectively supplied as pan control signals from the sound source position estimation unit 70 described later.
The imaging direction control means of the present invention corresponds to the motor control circuit of the imaging units 60a and 60b in the present embodiment.
The motor control circuit performs zoom control of the imaging units 60a and 60b based on S86a and S86b supplied as zoom control signals from the image recognition unit 80 described later.

Sound source position estimation unit 70
The sound source position estimating unit 70 estimates the sound source position based on the sound source direction microphone numbers MC_a and MC_b (signals S57a and S57b) supplied from the sound source direction detecting units 50a and 50b. The intersection in the direction in which the sound source direction microphone numbers MC_a and MC_b are respectively directed is uniquely determined on the horizontal plane of the room R1 shown in FIG. 24, and the intersection is estimated as the sound source position.

  The sound source direction detection units 50a and 50b select the sound source direction microphone numbers MC_a and MC_b by comparing the sound pressure levels of the sound signals input from the plurality of microphones, as described above. Since the microphone number is finally determined by the method, the sound source direction includes a slight error. However, as will be described later, since the zoom control of the imaging units 60a and 60b is performed based on the sound source position estimated by the sound source position estimating unit 70, the subject is adjusted to be captured in an enlarged manner. There is no problem even if the estimation of the sound source position performed by the estimation unit 70 includes some errors.

The sound source position estimation unit 70 has a built-in rewritable nonvolatile memory. In this memory, position information arranged on the horizontal plane of the sound source direction detection units 50a and 50b and the imaging units 60a and 60b in the room R1 is stored in advance, and the sound source position estimation unit 70 stores the calculated sound source position and the sound source position. Then, geometric calculation is performed from the position information of the imaging units 60a and 60b stored in the memory, and the rotation angles necessary for the imaging units 60a and 60b to capture the sound source position are calculated.
The sound source position estimation unit 70 outputs pan control signals S76a and S76b including rotation angles necessary for the image capturing units 60a and 60b to capture the sound source position to the image capturing units 60a and 60b, respectively.

Image recognition unit 80
The image recognition unit 80 sequentially inputs the moving image data captured by the imaging units 60a and 60b as signals S68a and S68b shown in FIG. Actually, the data acquired by the imaging units 60a and 60b is sequentially transferred to the buffer in the image recognition unit 80 in units of frames, and the image recognition unit 80 holds the data in units of frames as moving image data.
Note that display means (for example, an LCD or the like) for displaying images taken by the imaging units 60a and S60b from outside the room R1 to be imaged is incorporated. This display means corresponds to the display means of the present invention.
Further, the display means may be installed outside that is electrically connected to the image recognition unit 80.

In addition, the image recognition unit 80 performs an image recognition process in order to identify a person who is a subject from the video data displayed on the display unit.
As the image recognition processing, various known techniques are already known, and these known techniques can be applied. In general, the following processing is performed in the image recognition processing.

That is, a correlation coefficient between a part of an image to be recognized and a template (search shape) is calculated, and a position of a part of the image having a feature close to the template is specified based on the correlation coefficient.
Specifically, the following steps are executed.
a. Perform image normalization.
b. Coarse search for multiple locations that most likely match
c. A correlation coefficient is calculated for a plurality of searched positions.
d. Narrow down the position with the highest correlation.
At this time, focusing on the contour of the image recognition object and extracting the contour feature of the template image, the contour is registered as a model. Then, the above correlation coefficient is calculated from the contour feature of the template and the contour feature of the target image. That is, a calculation is made as to how similar the two contours are.
Through the above steps, the position of the image most likely to match the template is specified.

  In this embodiment, the image recognition unit 80 holds a template (contour feature) corresponding to a human face, and searches for the contour of the human face appearing in the moving image data acquired from the imaging units 60a and 60b based on this template. (Search for. Then, through the above-described image recognition processing, the image recognition unit 80 can recognize which position on the screen of the imaging units 60a and 60b is the position of the human face to be searched (searched).

  When recognizing the position of the face on the screen of the imaging units 60a and 60b, the image recognition unit 80 captures the imaging target 60a and 60b so that the face to be image-recognized is largely captured in the central area of the screen. The zoom control amount of each of 60b is calculated and output to the motor control circuit of the imaging units 60a and 60b as zoom control signals S86a and S86b.

In the above, each component of the imaging device 40 in this embodiment has been described.
Next, the operation of the imaging device 40 will be described with reference to FIGS.

Operation of Imaging Device FIG. 25 is a diagram for explaining the outline of the processing operation of the imaging device 40 in the present embodiment.
FIG. 25 illustrates an operation in which an image of the object OBJ1 is captured by the imaging units 60a and 60b when the object OBJ1 existing in the room R1 to be imaged emits sound. When the object OBJ1 emits sound, the sound source direction is first specified by the sound source direction detection units 50a and 50b. That is, the sound source direction detection units 50a and 50b specify that the direction of the lines L50a and L50b shown in FIG. 25 is the sound source direction.
In FIG. 25, the direction indicated by the lines L50a and L50b does not coincide with the direction in which the microphones of the sound source direction detection units 50a and 50b are directed, but FIG. 25 is a diagram for convenience of explanation, and the sound source direction detection unit 50a. , 50b, if the number of microphones is more than six, the resolution in the direction of the sound source to be specified can be improved, and both can be matched. Here, description will be made assuming that the positions of the sound source direction detection units 50a and 50b and the number of microphones thereof are appropriately set and the sound source detection direction matches the direction indicated by the lines L50a and L50b.

FIG. 26 is an example of a flowchart of main processing of the imaging apparatus 40 in the present embodiment. The main process is executed by controlling the sound source position estimation unit 70 and the image recognition unit 80 by a CPU (not shown in FIG. 24).
In FIG. 26, when a sound is detected, a series of processing is started (step ST10).
That is, when a sound is detected, it is checked whether or not the sound source direction specified by the sound source direction detection units 50a and 50b that detected the sound has changed compared to the previous detection (step ST11).
If the sound source direction specified by each of the sound source direction detection units 50a and 50b is the same as the direction specified last time, it is considered that the object OBJ1 is sounding at the same position as the previous time, and thus the rotation of the imaging units 60a and 60b. No control is necessary and the process ends.
If the sound source direction specified by each of the sound source direction detection units 50a and 50b is different from the direction specified last time, it is considered that the object OBJ1 is sounding at a position different from the previous time, so that the sound source position estimation process (step ST12). ) And image recognition processing (step ST13).

FIG. 27 is a flowchart showing the sound source position estimation process of the imaging apparatus 40 in this embodiment, and is called at step ST12 of the main process described with reference to FIG.
As shown in FIG. 27, first, when it is confirmed that there is a change in the sound source direction detection data (sound source direction microphone number) of the sound source direction detection units 50a and 50b (step ST20), the detection data (sound source) of the two sound source direction detection units The intersection is calculated based on the direction microphone number (step ST21).
That is, since the sound source position estimating unit 70 stores the position information (coordinate information) where the sound source direction detecting unit 50a is installed, the position (sound source position) of the object OBJ1 shown in FIG. 25 based on this position information. Is calculated as the intersection of the lines L50a and L50b (step ST22).
Furthermore, the sound source position estimation unit 70 outputs pan control signals S76a and S76b including rotation angles necessary for the image capturing units 60a and 60b to capture the sound source position to the image capturing units 60a and 60b, respectively.
In response to the pan control signals S76a and S76b, the motor control circuits of the image capturing units 60a and 60b rotate the image capturing units 60a and 60b in the horizontal direction (pan) and perform motors so that the sound source position is the object to be imaged. Is controlled (step ST23).
Accordingly, as illustrated in FIG. 25, the imaging units 60a and 60b are rotated in the directions indicated by the lines L60a and L60b, respectively, and the object OBJ1 is captured as a subject.

FIG. 28 is a flowchart showing an image recognition process of the imaging apparatus 40 in the present embodiment, and is called at step ST13 of the main process described with reference to FIG.
When the image pickup units 60a and 60b are rotated so that the sound source position becomes a subject to be photographed by the sound source position estimation process, the image recognition unit 80 first checks whether there is a video output (step ST30). Then, the contour of the object OBJ1 is searched according to the image recognition processing algorithm (step ST31).
For example, when a human face is registered as a template for image recognition, the correlation between a part of the contour feature of the image data and the registered contour feature of the face is calculated, and the image data having a high correlation Is recognized as a human face.
Then, the image recognition unit 80 calculates the zoom control amount so that the portion of the image data recognized as the face of the subject is displayed on the screen, and images the zoom control signals S86a and S86b including the calculated zoom control amount. It is sent to the units 60a and 60b (step ST32).
The motor control circuits of the imaging units 60a and 60b that have received the zoom control signals S86a and S86b perform zoom control of the imaging units 60a and 60b, respectively. As a result, for example, the face of the subject is displayed large on the screen.

  As described above, according to the imaging apparatus of the present embodiment, a sound source based on a plurality of sound source direction detection units that specify a sound source direction using a plurality of microphones and a microphone number specified by the plurality of sound source direction detection units. A sound source position estimation unit that estimates a position, an imaging unit that rotates with respect to the sound source position estimated by the sound source position estimation unit, and shoots a moving image, and performs image recognition processing based on the moving image data obtained by the imaging unit When the subject is specified, the image recognition unit 80 that controls the imaging unit so that the subject is enlarged and captured is provided, and the following effects can be obtained.

  That is, since the position of an object (subject) including a human being as a sound source can be grasped, the subject can be clearly photographed by a relatively small number of imaging means, and the system as an entire imaging apparatus is provided. Cost can be suppressed.

  When the imaging means are arranged on both sides of the room to be imaged, it is possible to shoot from two directions opposite to the subject. Therefore, when the object is a human being, the physical characteristics such as the face are more improved. It becomes possible to grasp clearly.

  It should be noted that various embodiments can be changed without departing from the content of the above-described embodiment and without changing the gist of the present invention.

  For example, in the above embodiment, the sound source direction detection units 50a and 50b each have six directional microphones, but the sound source direction detection unit has at least two directional microphones having different sound collection directions. If there are at least two and at least two sound source direction detection units located at positions apart from each other, the sound source position is uniquely determined geometrically. Therefore, as in the above-described embodiment, six directional microphones are not necessarily provided. It is not necessary to arrange them evenly.

However, since the accuracy of the estimated sound source position is improved as the number of directional microphones included in the sound source direction detection unit is increased, it can be said that the larger the number of directional microphones, the better. However, if the monitoring direction is limited, the accuracy of sound source position estimation can be improved to some extent even with a small number of microphones.
For example, as clearly shown in FIG. 24, the side where the sound source position estimation unit 70 of the sound source direction detection units 50a and 50b is arranged has a very small monitoring area, so all six directional microphones are on the imaging unit 60b side. If it is arranged so as to point to the sound source, the estimation accuracy of the sound source position can be improved without increasing the number of microphones.

  In the imaging apparatus of the above embodiment, the two sound source direction detection units are configured. However, the number is not limited to two, and for example, by using three or four sound source direction detection units, sound source position estimation is performed. The accuracy of the sound source position calculated by the unit 70 is improved, and at the same time, a wider range of the room R1 can be captured by the imaging unit. As a result, a so-called blind spot in photographing is reduced.

  In addition, when there are three or more sound source direction detection units, it is possible to detect the sound source position in three dimensions. In this case, a tilt (up and down rotation) function is further provided for the imaging unit. By adding, it becomes possible to perform photographing at an arbitrary three-dimensional position in the room to be imaged.

  In addition, the image recognition unit 80 calculates the zoom control amount of each of the image capturing units 60a and 60b so that the image capturing units 60a and 60b capture a large face in the center area of the screen, and the zoom control signal S86a. , S86b is output to the motor control circuit of the imaging units 60a, 60b. However, instead of such feedforward control, a control system using a feedback servo may be configured. That is, the difference between the position of the face on the screen and the target value (the position of the face that should be largely projected in the center of the screen) is fed back from the image recognition unit 80 to the imaging units 60a and 60b, and the motor control of the imaging units 60a and 60b is performed. The circuit may be configured to determine the zoom control amount.

  In the embodiment described above, the image recognition unit 80 calculates a necessary zoom control amount for the imaging units 60a and 60b after specifying an object such as a face by image recognition processing. Is specified by the sound source position estimation unit 70, and the necessary zoom control amount can be calculated from the coordinates of the sound source position. In this case, an accurate zoom control amount corresponding to the size of the object cannot be calculated as in the case where the image recognition process is executed. However, since the image recognition process is not required, the zoom control response is fast and the system There is an advantage that it is inexpensive as a whole.

(A) is a figure which shows the outline | summary of the conference system as an example to which a two-way communication apparatus is applied, (B) is a figure which shows the state in which the telephone apparatus in (A) is mounted, (C) is a diagram showing the arrangement of the communication device placed on the table and the conference participants. It is a perspective view of a two-way communication device. FIG. 2 is an internal sectional view of the two-way communication device illustrated in FIG. 1. FIG. 2 is a plan view of a microphone / electronic circuit housing unit from which an upper cover of the two-way communication device illustrated in FIG. 1 is removed. It is a figure which shows the connection state of the main circuit of a microphone and an electronic circuit accommodating part, and has shown the connection state of the connection of a 1st digital signal processor (DSP1) and a 2nd digital signal processor (DSP2). FIG. 5 is a characteristic diagram of the microphone illustrated in FIG. 4. (A)-(D) is a graph which shows the result of having analyzed the directivity of the microphone which has the characteristic illustrated in FIG. It is a partial block diagram of the deformation | transformation aspect of a two-way communication apparatus. It is a graph which shows the outline | summary of the whole processing content in a 1st digital signal processor (DSP1). It is a flowchart which shows the 1st form of the noise measuring method of a two-way communication apparatus. It is a flowchart which shows the 2nd form of the noise measuring method of a two-way communication apparatus. It is a flowchart which shows the 3rd form of the noise measuring method of a two-way communication apparatus. It is a flowchart which shows the 4th form of the noise measuring method of a two-way communication apparatus. It is a flowchart which shows the 5th form of the noise measuring method of a two-way communication apparatus. It is drawing which shows the filtering process in the telephone apparatus of this invention. It is a frequency characteristic figure which shows the process result of FIG. It is a block diagram which shows a band pass filtering process and a level conversion process. It is a flowchart which shows the process of FIG. It is a graph which shows the process which determines the speech start and completion | finish of a two-way communication apparatus. It is a graph which shows the flow of the normal process of a two-way communication apparatus. It is a flowchart which shows the flow of the normal process of a two-way communication apparatus. It is the block diagram which illustrated the microphone switching process of a two-way communication apparatus. It is the block diagram which illustrated the method of the microphone switching process of a two-way communication apparatus. It is a figure which shows an example of a structure of the imaging device which is embodiment of this invention. It is a figure which shows the principle of operation of the imaging device which is embodiment of this invention. It is a flowchart which shows the processing operation of the imaging device which is embodiment of this invention. It is a flowchart which shows the processing operation of the sound source position estimation process of the imaging device which is embodiment of this invention. It is a flowchart which shows the processing operation | movement of the image recognition process of the imaging device which is embodiment of this invention.

Explanation of symbols

  DESCRIPTION OF SYMBOLS 1 ... Two-way communication apparatus, MC1-MC6 ... Microphone, 16 ... Reception speaker, 23 ... Microprocessor, 24 ... Codec, 25 ... 1st digital signal processor (DSP1), 26 ... 2nd digital signal processor (DSP2) 27 ... A / D converter block, 28 ... D / A converter block, 29 ... amplifier block, 40 ... imaging device, 50a, 50b ... sound source direction detection unit, 60a, 60b ... imaging unit, 70 ... sound source position estimation 80, an image recognition unit.

Claims (7)

Two or more microphone selecting means for arranging at least two microphones having directivity so as to be directed in different sound collecting directions, and selecting one microphone based on sound pressure collected by the microphone;
A sound source position specifying device comprising sound source position specifying means for specifying a sound source position based on a direction in which at least two microphones are directed among a plurality of microphones selected by the two or more microphone selection means. Two or more microphone selecting means for arranging at least two microphones having directivity so as to be directed in different sound collecting directions, and selecting one microphone based on sound pressure collected by the microphone;
Sound source position specifying means for specifying a sound source position based on a direction in which at least two microphones are directed among a plurality of microphones respectively selected by the two or more microphone selection means;
Imaging means for changing the imaging direction by rotation;
An imaging apparatus comprising: an imaging direction control unit that controls the imaging direction by rotating the imaging unit so that the imaging unit images the sound source position specified by the sound source position specifying unit. Display means for displaying the video obtained by the imaging means;
The imaging apparatus according to claim 2, wherein the sound source position specified by the sound source specifying position is enlarged and displayed on the display unit. Image recognition means for recognizing a specific target from the video displayed by the display means,
The imaging apparatus according to claim 3, wherein the specific object recognized by the image recognition means is enlarged and displayed on the display means. At least two microphones having directivity are arranged so as to be directed in different sound collection directions, and rotated with two or more microphone selection units that select one microphone based on the sound pressure collected by the microphone. And a monitoring method having an imaging unit that changes the imaging direction by:
Identifying a sound source position based on a direction in which at least two microphones are directed among a plurality of microphones respectively selected by the two or more microphone selection units;
Rotating the imaging unit so that the imaging unit images the identified sound source position. The imaging method according to claim 5, further comprising a step of enlarging and displaying the sound source position imaged by the imaging unit. Displaying an image captured by the imaging unit;
Recognizing a specific target from the displayed video;
The imaging method according to claim 5, further comprising: enlarging and displaying the recognized specific object.

Images