Patent Publication Number: US-2006002686-A1

Title: Reproducing method, apparatus, and computer-readable recording medium

Description:
BACKGROUND OF THE INVENTION  
      The present invention relates to a reproducing method, an apparatus and a computer-readable recording medium for naturally reproducing the image information and the voice information, which are inputted through a network having a heavy traffic load.  
      In recent years, there has spread the network system, in which an image is taken by a network camera and is transmitted to a computer system through the network such as the internet. However, this network system can acquire the image information by controlling the computer system but not the surrounding voice information. Thus, there has been developed the network camera (as will be called the “voice mapping type network camera”), which is enabled to perform not only the image communication but also the voice communication by mounting a speaker and a microphone.  
       FIG. 8  is an explanatory diagram of a network system for the voice communication of the related art. In this network system, in connection with the transmission of an image, an image taken by a camera  1  of a voice mapping type network camera  1  is compressed by an image processor  12 , and this compressed image data is processed in a protocol by a communication control unit  13 . This processed data is sent to a network  3  and to a computer system  2 . This computer system  2  decompresses the image data received, and displays it on a screen.  
      On the other hand, the image to be taken is processed into an image of a desired angle and zoom by controlling the pan, tilt and zoom   camera by the (not-shown) camera control unit. The browser (i.e., the pe  program of screen displaying information) of the computer system  2  displays the portal screen showing the image and the control bar in the monitor, when it receives the portal screen displaying information through the network  3 . When the user controls the pan, tilt and zoom by the control bar, the JAVA (of a registered trade mark) applet or the like transmits the IP packet confining data of a control quantity from the communication control unit  13  to the voice mapping type network camera  1 . In this voice mapping type network camera  1 , a control unit  9  extracts the data from that IP packet and transmits the control quantity to the camera control unit so that the (not-shown) pan motor, the (not-shown) tilt motor and the (not-shown) linear actuator are driven to change the taking direction and the zoom of the camera  10 .  
      Next, in connection with the voice communication, the voice inputted from a microphone  17  is subjected to an AD conversion and a compression by a voice transmission processor  15  so that the voice transmission data is sent through the communication control unit  13  and the network  3  to the computer system  2 . This computer system  2  processes the voice transmission data received, and outputs a voice from a speaker  28 . Likewise, the voice inputted from a microphone  27  of the computer system  2  is processed by the computer system  2  and is transmitted as the voice reception data so that it is sent through the network  3  to the voice mapping type network camera  1 . In this voice mapping type network camera  1 , the voice reception data received is transferred through the communication control unit  13  to a voice reception processor  14 , in which the data is decompressed and DA-converted and is outputted to a speaker  18 .  
      In case this voice mapping type network camera  1  transmits the image and the voice to the computer system  2 , a time stamp is generally made on the individual data of the image and the voice, that is, the transmission is made by adding synchronous information of time information (as referred to JP-A-9-27871, for example). Both the voice and image data are given the synchronizing information by the time control, and the data having the synchronizing information is reproduced on the reception side so that both the voice and image data are synchronously outputted. At this time, the voice has a determined data length, but the image data is not determined on its output time. In case the network has a heavy traffic load, therefore, this terminal device finds it difficult to transmit all the image data and the voice data, and thins the data. As a result, the image and the voice are partially cut to interrupt the voice. The interrupted voice is hard to listen thereby to deteriorate the information transmission seriously.  
      There also exists the time stamp method, in which a synchronization is made by adding a frame number to the image data and the voice data. However, the time stamp and the frame number have to be added individually to the image data and the voice data. In case the configuration is so complicated that the network has a heavy traffic load, it is difficult for the terminal device to transmit all the image data and the voice data. As a result, the voice is interrupted, and the configuration is complicated to raise the cost.  
      There has also been proposed a multimedia multiplex transmission device, which does not cut the voice but creates a multiplex signal efficiently in case the voice signal is a voiceless sound (JP-A-2001-16263). This device is provided with a voice signal buffer unit and a voiceless sound detecting unit, and the voice signal buffer unit stores the voiced encoded signal temporarily. When there is detected the case in which the voice signal caught by an external microphone is voiceless, the write of the data is enabled, in case the input signal from the voiceless sound detecting unit is at a low level, but disabled in case the same is at a high level. Thus, the time area assigned to the voice signal of the multiplex signal is not uselessly assigned to the vide-encoded signals. In case a voice is made voiceless in the processing, a time period longer than necessary is taken from a low level to a high level. In case a voiceless sound is changed into a voiced sound, the level is instantly changed from high to low. These changes do not break the voices at the head and end of a word.  
      The voice mapping type network camera of JP-A-9-27871 transmits image and voice data, a synchronization is made by adding synchronous information such as time information to each image and voice data or by adding a frame number to each image and voice data. In the case of a heavy traffic load of the network, however, those synchronizing methods have found it difficult to transmit all the image data and the voice data. If a delay occurs, the data have to be thinned so that the image and voice reproduced are partially cut and interrupted. Moreover, these techniques are just to thin the data on the data transmission side but are not solutions for the problem on the reception side, which receives the influences of the traffic fluctuations. If the traffic load is heavy, the packet of the voice data is delayed not to decrease but to increase the voice delay in the voice buffer of the computer system.  
      On the other hand, the multimedia multiplex transmission device of JP-A-2001-16263 includes a voice signal buffer unit and a voice/voiceless detection unit. In case the voice signal detected by the outside microphone is voiceless, the device does not cut the voice but inhibits the data write so that the device can create the multiplex signal efficiently. In case the voice signal of the external microphone is a voiceless sound, the area assigned to the voiceless sound signal of the multiplex signal to be sent from the multimedia multiplex transmission device is assigned to the video encoding signal. Therefore, this technique does not solve either the problem of the computer device on the reception side. The problem thus far described is left unsolved if the traffic load is heavy.  
     SUMMARY OF THE INVENTION  
      In view of the above problems of the related art, therefore, the invention has an object to provide an apparatus, a method for reproducing image information and voice information, and a computer-readable recording medium, which can utilize a buffer effectively even with much no-sound data or with a delayed packet.  
      In order to achieve the above object, according to the present invention, there is provided a terminal for storing voice information, if received through a network, temporarily in a voice reception buffer, for decoding the voice information outputted from the voice reception buffer and for outputting a voice after DA-converted. This terminal includes a buffer control unit for controlling the input/output of the voice information to and from the voice reception buffer, and a reception buffer level determining unit for deciding it as a no-data or a no-sound that the voice information in the voice reception buffer is at a predetermined peak value or less continuously for a predetermined time period, and it as a sound that the same exceeds the peak value. The terminal is mainly characterized in that the voice information determined as the no-data or no-sound by the buffer control unit is discarded, and in that the remaining voice information is compacted and outputted to a voice processing unit.  
      According to the apparatus, the method for reproducing image information and voice information, and the computer-readable recording medium of the invention, the delay is improved by discarding the voiceless portion even if the voice delay increases. 
    
    
     BRIEF DESCRIPTION OF THE DRAWINGS  
      The above objects and advantages of the present invention will become more apparent by describing in detail preferred exemplary embodiments thereof with reference to the accompanying drawings, wherein:  
       FIG. 1A  is a configuration diagram of a network camera according to a first embodiment of the invention, and  FIG. 1B  is an internal block configuration diagram of the inside of a control unit of the network camera according to the first embodiment of the invention;  
       FIG. 2  is a block configuration diagram of a computer system according to the first embodiment of the invention;  
       FIG. 3A  is an explanatory diagram of a portal screen display of the computer system according to the first embodiment of the invention, and  FIG. 3B  is an explanatory diagram of a setting screen for a no-sound erasure of  FIG. 3A ;  
       FIGS. 4A  to  4 E are explanatory diagrams of the data processing of a voice reception buffer of the computer system according to the first embodiment of the invention;  
       FIG. 5  is an explanatory diagram of a data discard of a voice reception buffer according to the first embodiment of the invention;  
       FIG. 6  is an explanatory diagram of threshold settings for deciding no-data and no-sound of the voice reception buffer according to the first embodiment of the invention;  
       FIG. 7  is a flow chart at the time when the no-data and no-sound data are discarded by the network camera and the computer system according to the first embodiment of the invention;  
       FIG. 8  is an explanatory diagram of a list of display of images for the voice communications of the related art;  
       FIG. 9  shows a hardware configuration diagram of a camera according to a second embodiment of the invention; and  
       FIG. 10  shows an appearance of the camera according to the first embodiment of the invention. 
    
    
     DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS  
      In order to solve the above-specified object, according to the invention, there is provided a method for reproducing and outputting image information and voice information by receiving the image information and the voice information from a camera through a network, comprising: storing said voice information; deciding it as a no-data or a no-sound that said voice information is lower than a predetermined threshold, and it as a voice that the same is higher than a predetermined threshold; and discarding the voice information decided as the no-data or the no-sound, and compacting the remaining voice information.  
      The voice information decided as the no-data or the no-sound in the voice reception buffer is discarded, and the remaining voice information is compacted and outputted as a voice. As a result, the voice reception buffer can be effectively utilized, and the voice is neither delayed from the image nor cut. Thus, the method of the invention is hardly influenced by the traffic fluctuations.  
     DESCRIPTION OF THE PREFERRED EMBODIMENTS  
     First Embodiment  
      A network camera, a program and a network system according to the first embodiment of the invention will be described in the following. As to  FIGS. 1A through 6 , the same reference characters as those of a voice mapping type network camera  1  and a computer system  2  of  FIGS. 7 and 8  are basically unchanged in the first embodiment.  
      In  FIGS. 1A and 1B , numeral  1  designates a voice mapping type network camera (or a network camera of the invention) having such a voice communication device mounted thereon and capable of taking and sending image and having voice communications, numeral  2  designates a computer system (or a terminal of the invention) such as a personal computer capable of having the voice communications, and numeral  3  designates a network such as the internet or Ethernet (of a registered trade mark). Numeral  10  designates a camera of the voice mapping type network camera  1 , and numeral  10   a  designates a camera control unit for controlling the pan, tilt and zoom of the camera  10 . Numeral  10   b  designates a pan motor for controlling the panning action of the camera  10 ; numeral  10   c  a tilt motor for controlling the tilting action of the camera  10 ; and numeral  10   d  a linear actuator for a feeding action to control the zooming of the camera  10 .. The computer system  2 A controls the panning, tiling and zooming operations based on a control bar on the portal screen. The control bar is acquired from the voice mapping type network camera  1  and displayed on the computer system  2 A. Then, the IP packet containing the data of the panning, tilting and zooming control variables is transmitted from the computer system  2  by the JAVA (of the registered trade mark) applets. The voice mapping type network camera  1  extracts the control data from that IP packet and transmits the control variables to the camera control unit  10   a  so that the pan motor  10   b , the tilt motor  10   c  and the linear actuator  10   d  are individually driven to change the imaging direction and the zoom.  
      Numeral  11  designates a codec unit for compressing and decompressing the data to be transmitted and received. Numeral  12  designates an image processor for compressing the image signals taken by the camera  10 . Numeral  13  designates a communication control unit for processing the image data compressed by the image processor  12  in the protocol and for transmitting the processed data. Here, this protocol processing indicates the processing such as the TCP/IP protocol or the IEEE 802.03 protocol of the Ethernet (of the registered trade mark).  
      Numeral  14  designates a voice reception processor for decoding the voice reception data (or the PCM data) received by the voice mapping type network camera  1 . Numeral  14   a  designates a DA converter for converting the output or a digital signal of the voice reception processor  14  into an analog signal. Numeral  15  designates a voice transmission processor for encoding the voice inputted to the voice mapping type network camera  1 . Numeral  15   a  designates an AD converter for converting the output or an analog signal from a voice input adjusting circuit  17   a  (as will be described hereinafter). Numeral  16  designates a buffer unit of the voice mapping type network camera  1 . Numeral  16   a  designates an image buffer of the buffer unit  16  and for the image data such as JPEG or MPEG compressed by the image processor  12 . Numeral  16   b  designates a voice transmission buffer of the buffer unit  16  and for the PCM data encoded by the voice transmission processor  15 . Numeral  16   c  designates an FIFO (first In First Out) voice reception buffer of the buffer unit  16  and for buffering the PCM data transmitted from the computer system  2  via the network  3 .  
      This voice reception buffer  16   c  buffers a large quantity of voice reception data transmitted, temporarily in accordance with the relationship between the ability and quantity of processing. In a case that the traffic load rises, the data to arrive is decreased due to the delay of packets so that the processing seems to have no problem. However, a problem is occurred that the time band in which the data cannot be fetched is continues so that a no-data area is mixed into the data of the voice reception buffer  16   c . Specifically, the first-in data is continuously outputted, but the data of packet delay is not written in the memory elements configuring the voice reception buffer  16   c , that is, the memory elements are not charged. When this state of no data is transferred to the voice reception processor  14 , this voice reception processor  14  has to perform the meaningless processing. In the First embodiment, therefore, this area of no data and the intrinsic no-sound state of a low volume of sound are detected and discarded. The no-data and the no-sound will be called together the “no-data/no-sound”.  
      Next, in  FIG. 1A , numeral  17  designates a microphone for inputting the voice around the voice mapping type network camera  1 , numeral  18  designates a speaker for outputting a voice, and numeral  18   a  designates a voice output adjusting circuit. The (not-shown) echo cancellers may be interposed between the microphone  17  and the voice transmission processor  15 , and between the speaker  18  and the voice reception processor  14  to prevent such a loop for creating an echo from being established that the voice outputted from the speaker  18  is inputted again to the microphone  17  and outputted from a speaker  28  on the side of the computer system  2  and is again inputted from a microphone  27 .  
      In  FIGS. 1A and 1B , numeral  19  designates a control unit of the voice mapping type network camera  1 , numeral  19   a  designates a communication execution unit (or the communication unit of the invention) for performing a voice communication and an image transmission when the voice communication mode is selected from the computer system  2 ; and numeral  19   b  designates an image displaying information generation unit for creating the screen displaying information to be transmitted. from the voice mapping type network camera  1  to the computer system  2 . Numeral  19   c  designates a flag for indicating the communication state of the plural computer systems  2  accessing to the voice mapping type network camera  1 , such as the state in which the voice is being transmitted, in which the voice is being received, or in which the right to control the pan, tilt or zoom is being exercised, and numeral  19   d  designates a file transfer unit for downloading the program to control the computer system  2 , i.e., the later-described terminal side communication processing unit  26 , such as the program of the active x or the JAVA (of the registered trade mark) applets stored in a transmission file storage  20   b.    
      Next, numeral  19   e  designates a buffer control unit for controlling the write action and the output action of the PCM data in and to the voice reception buffer unit  16   c . Numeral  19   f  designates a reception buffer level decision unit for deciding whether or not the level of the data corresponds to the no-data/no-sound; and numeral  19   g  designates a timer unit for counting whether or not the state of no-data/no-sound continues for a predetermined time  10 . period. In the first embodiment, in case the no-data/no-sound continues for the predetermined time period, the buffer control unit  19   e  discards (or erases the charge) the entire data of the time period and makes the control to eliminate the area of the no-data/no-sound by advancing the discarded area to the subsequent data. The reception buffer level decision unit  19   f  is set with a threshold for evaluating the voice and the no-data/no-sound. The reception buffer level decision unit  1   9   f  decides the no-data/no-sound, when the level is at the threshold or lower, and informs the buffer control unit  19   e  of the decision. In the first embodiment, the no-data/no-sound is decided when the detected level is at the threshold or lower for 365 ms, but a proper set value may be adopted for the time duration. In response to this notice, the buffer control unit  19   e  causes the timer unit  19   g  to count a predetermined time, so as to decide whether or not the no-data/no-sound continues. When the timer unit  19   g  counts out, it is decided that the no-data/no-sound has occurred. Moreover, numeral  19   h  designates a setting unit for setting the aforementioned threshold. Next in  FIG. 1A , numeral  20  designates a storage unit stored with program or the like for controlling the system, numeral  20   a  designates a screen displaying information storage stored with the template of portal screen displaying information or other screen displaying information (e.g., web pages), and numeral  20   b  designates a transmission file storage stored with the program (as will be called the “terminal side communication processing unit”) to be transmitted to the computer system  2  and executed by the CPU of the computer system  2 , such as the active x or the JAVA (of the registered trade mark). Numeral  20   c  designates an image storage for storing the image data compressed in the image processor  12 . Here, the above screen displaying information described with the HTML or the like is stored in the screen displaying information storage  20   a . When, however, the images of the individual voice mapping type network camera  1  are to be displayed in a list with the portal screen displaying information, each image data displayed is stored in the image storage  20   c  of the voice mapping type network camera  1 .  
      The configuration of the computer system  2  will be described with reference to  FIG. 2 . In  FIG. 2 , numeral  21  designates a communication control unit acting as an interface with the network  3 , numeral  22  designates a control operation unit equipped with a CPU as a hardware and realized as a function realizing unit by reading the program from a storage unit  23  that is configured to store a program and data, and numeral  23   a  designates a voice reception buffer for storing voice data. Moreover, numeral  24  designates a browser unit for acquiring and perusing the image displaying information from the web site on the network  3 , and numeral  25  designates a voice processing unit realized as a function realizing unit with the voice processing program such as the JAVA (of the registered trade mark) applet program or the plug-in.  
      Moreover: numeral  25   a  designates buffer control unit for controlling the write action and the output action of the PCM data to the voice reception buffer  23   a , numeral  25   b  designates a reception buffer level decision unit for decision whether or not the level is equivalent to the no-data/no-sound; and numeral  25   c  designates a timer unit for counting whether or not the state of no-data/no-sound continues for a predetermined time period. Moreover, numeral  25   d  designates a screen displaying information generation unit for creating a no-sound erasure setting screen  56  (as referred to  FIG. 3B ) to vary the threshold for deciding the no-data/no-sound at the voice reception buffer  23   a , with the buffering data length. Moreover, numeral  25   e  designates a setting unit for setting the threshold when the buffering data length is inputted from the no-sound erasure setting unit  56 .  
      On the other hand, the numeral  26  designates the terminal side communication processing unit which is realized by the program downloaded by the file transfer unit  193  of the voice mapping type network camera  1 , such as the active x or the JAVA (of the registered trade mark). The numeral  27  designates the microphone, numeral  27   a  designates a voice input adjusting circuit, the numeral  28  designates the speaker, numeral  28   a  designates a voice output adjusting circuit, numeral  29  designates a display unit, and numeral  30  designates a motor.  
      With reference to  FIGS. 3A and 3B , here will be described the portal screen displaying information and the no-sound erasure setting screen, which are transmitted to the computer system  2  by the voice mapping type network camera  1  of the first embodiment. In  FIG. 3A , numeral  51  designates an image area of time-varying images or still images, and numeral  52  designates a control bar for controlling the pan, tilt and zoom of the camera  10  of the voice mapping type network camera  1 . Numeral  52   a  designates a direction control button, and numeral  52   b  designates a zoom adjusting bar. Here, the control bar  52  is prepared with a button for calling a set screen to discard the later-described no-data/no-sound data. Numeral  53  designates a voice transmission button for transmitting the voice to the voice mapping type network camera  1  when depressed, and numeral  54  designates a voice reception button for receiving the voice made in the voice mapping type network camera  1 . Numeral  55  designates a volume adjusting bar for adjusting the volume to be outputted from the speaker  18  of the voice mapping type network camera  1 . The client of the voice mapping type network camera  1  receives and displays the portal screen displaying information on the monitor  30 , and controls the direction control button  52   a  and the zoom adjusting bar  52   b , while observing the images on the portal screen, to switch the angle or the like of the camera  10  thereby to acquire a new image. In the voice transmission mode, on the other hand, the client pushes the voice transmission button  53  to transmit the voice, and pushes the voice reception button  54  to receive the voice on the side of the voice mapping type network camera  1 .  
      Subsequently in  FIG. 3B , the numeral  56  designates the no-sound erasure setting screen for varying the threshold to decide the no-data/no-sound in the voice reception buffer  23   a , as described above, with the data length, and numeral  57  designates a setting box for setting the buffering data length. For simplicity, this screen will be called the “no-sound erasure setting screen”. When the no-sound erasure setting button displayed in the control bar  52  of the portal screen is pushed, the no-sound erasure setting screen  56  created by the displaying information generation unit  25   d  is called and displayed in the monitor  30 . Buffering data lengths such as 400 ms, 500 ms, 600 ms, 700 ms, 800 ms, 900 ms and 1,000 ms are inputted to the setting box  57 , one of which can be selected, as shown in  FIG. 6 . Although detailed hereinafter, the threshold for deciding the no-data/no-sound may take one value. In  FIG. 6 , however, a pair of different thresholds are individually set for the case, in which the state is varied from the no-data/no-sound to the voice, and for the case, in which the state is varied from the voice to the no-data/no-sound. Specifically, the thresholds are set to the threshold H (dB) at the time when the state is varied from the no-data/no-sound to the voice, and to the threshold L (dB) at the time when the state is varied from the voice to the no-data/no-sound. When the buffering data length is inputted as 400 ms at the setting box  57 , for example, the threshold H is set to −9 dB whereas the threshold L is set to −12 dB by the setting unit  25   e.    
      Subsequently, the actions to discard the no-data/no-sound at the voice reception buffer  23   a  of the computer system  2  will be described in detail with reference to  FIGS. 4A  to  4 E,  FIG. 5  and  FIG. 6 .  FIG. 4A  shows the IP packet containing the voice data transmitted from the voice mapping type network camera  1 . The voice data of one frame is stored after the header. These voice data are fetched by the communication control unit  21 , and the buffer control unit  25   a  transfers the PCM data of 8 bits at the unit of 8 bits to a predetermined column of the voice reception buffer  23   a . In 8 bits of the PCM data, as shown in  FIG. 4B , the leading 1 bit is assigned to discriminate the polarity (+, −), and the remaining 7 bits indicate the peak value. Since the compression coefficients are different according to the so-called “μ rule” or “A rule”, so that the PCM data take different values according to the compression methods.  
      The buffer control unit  25   a  shown in  FIG. 4C  has a buffer capacity of (8 x n) bits in the FIFO and includes memory element arrays of n columns at the unit of 8 bits. The buffer control unit  25   a  transfers and writes the PCM data at the leading end, and outputs the PCM data at a predetermined rate of 8 bits at the terminal end so as to output the voice at a uniform rate. After outputting, the charges (indicating the PCM data) of the remaining columns are transferred sequentially for each column to the terminal end.  
      Here, the graph of  FIG. 4D  illustrates the peak values of the PCM signals. The data of k columns corresponding to the width of T ms (e.g., 365 ms in the first embodiment) become lower than the threshold L on the terminal side and higher than the threshold H on the leading side. Here, the peak values are the absolute values excepting the polarity (i.e., 1 bit). The PCM data of (8×k) bits for T ms have low peak values and are decided to be in the no-sound state so that they are discarded. In the case of no data, peak values 0 of a k number are arranged. The output is made at the unit of 8 bits, as shown in  FIG. 4E , and is inputted to the voice processing unit  25 . In this voice processing unit  25 , the input is converted into voice digital signals (or PAM signals), which are converted by the not-shown DA converter into analog signals and outputted as analog signals from the speaker  28 .  
      When data of a predetermined quantity is stored in the voice reception buffer  23   a , the buffer control unit  25   a  discards the no-data/no-sound data and sequentially compacts and outputs the voice data. The actions of the voice reception buffer  23   a  at this time will be described with reference to  FIG. 5 . In  FIG. 5 , the voice areas decided by the reception buffer level decision unit  25   b  are A, B and C, and the areas of the no-data/no-sound are M and N. The PCM signal becomes gradually smaller in the area A and than the threshold L at a point p, passes through the area M and crosses the threshold H at a point q so that it becomes the PCM signal in the area B. This PCM signal takes the maximum in the area B, crosses again the threshold L at the point p, passes through the area N and crosses the threshold H at the point q. Here, if the area A is positive, the area B is negative with an exception. Thus, the point p takes a lower threshold, and the point q takes a higher threshold. This difference is made to prevent the final data of the voice from being excessively cut. Moreover, the point p to be evaluated as the no-data/no-sound is set to the low value for ensuring the reliability. When the voice is returned, the area surely has passed through the area for the evaluation of the no-data/no-sound. Thus, the decision is not mistaken even if the threshold is made slightly higher at the point q.  
      The areas M and N of the no-data/no-sound thus decided are discarded (or discharged) by the buffer control unit  25   a , and the areas A, B and C are sequentially compacted. This state is shown in the two lower diagrams of  FIG. 5 . It is found that the buffer capacities are made sufficient. The areas A, B and C are continuous, and the output is made as if the no-data/no-sound are absent.  
      However, it cannot be said better that the decision of the no-data/no-sound is always made with the constant thresholds L and H. Specifically, the threshold H is lowered to increase the voice data for the voice, when the buffering data length of the voice reception buffer  23   a  is short. When the buffering data length is large, the threshold L and the threshold H are increased to reduce the voice data for the voice. These operations are preferred for causing no delay. With these decisions, moreover, the area of no data is always at the threshold L or lower. Even in case the threshold L and the threshold H are varied, it is possible to eliminate the influences due to the fluctuations in the traffic load of the network  3 .  
      In  FIG. 6 , the buffering data lengths can be set to 400 ms, 500 ms, 600 ms, 700 ms, 800 ms, 900 ms and 1,000 ms, and the threshold L and the threshold H are set with a hysteresis of 3 dB. With this difference of 3 dB, it is possible that the last data of the voice is not excessively cut, and it does not occur that the voice and the no-data/no-sound are misjudged.  
      The threshold L and the threshold H are increased in proportion to the buffering data length as this data length increases. This increase is reasoned in the following. In case the buffer capacity is large, it is frequently proportion to the size of the quantity of the PCM data received. If the range for deciding the no-data/no-sound is widened by raising the threshold L and the threshold H (or the thresh levels), it is possible to decrease the operations to be processed by the voice processing unit  25 . If the threshold H is set to −9 dB and the threshold L is set to −12 dB for the buffering data length of 400 ms, they can be preferably increased by 3 dB at every 100 ms from 400 ms to 1,000 ms so that the threshold H may take +9 dB whereas the threshold L may take +6 dB at 1,000 ms. The threshold L and the threshold H are varied with a difference of 3 dB for every 100 ms of the buffering data length.  
      Here, the description thus far made is directed mainly on the discard setting process and the erasing action of the no-sound data at the voice reception buffer  23   a  of the computer system  2 . Especially, the description is made on the computer system  2  for forming the voice reception buffer  23   a  by transmitting the program such as the JAVA (of the registered trade mark) applets from the voice mapping type network camera  1  and for configuring the terminal communication processor  26  to communicate, but the invention should not be limited thereto. Moreover, all these descriptions are similar to those of the discard setting process and the erasing actions of the no-sound data in the voice reception buffer  1   6   c  of the voice mapping type network camera  1 , and the detailed description is omitted because of the overlap. Here, the voice processing unit  25  of the computer system  2  performs the function of the voice reception processor  14  at the voice receiving time and the function of the voice transmission processor  15  at the voice transmitting time. In the computer system  2 , moreover, the client receives the portal screen and displays the no-sound erasure setting screen  56  for inputting the settings. In the case of the voice mapping type network camera  1 , however, the manager sets from the maintenance terminal.  
      Subsequently, here is described the flow for discarding the no-data/no-sound data between the network camera and the computer system according to the first embodiment of the invention.  FIG. 7  is a flow chart at the time when the no-data and no-sound data are discarded by the network camera and the computer system of the first embodiment. In  FIG. 7 , the routine is awaited (at step  1 ) till the voice data (or the PCM data) of a predetermined quantity is stored in the voice reception buffer  23   a . The reception buffer level decision unit  25   b  decides whether the voice data is the no-data/no-sound (at step  2 ) or valid-voice when the voice data is stored at a predetermined amount.  
      The reception buffer level decision unit  25   b  discards the voice data (at step  3 ) in the area of no-data/no-sound, and compacts the spaces of the voice areas sequentially (at step  4 ). The voice is inputted to the voice processing unit  25  and converted into the voice digital signals (i.e. the PAM signals) (at step  5 ) so that the analog signals are outputted (at step  6 ) from the speaker  28  through the DA converter.  
      Thus, the voice reception buffer  23   a  varies the buffering data length, and varies the thresh levels according to the quantity of the voice data stored. As a result, the quantity of processing of the voice processing unit  25  can be reduced according to the traffic state at the voice communication time. Even with much no-data and no-sound data or with a packet delay, the voice is neither delayed to make effective use of the buffer nor influenced by the traffic load.  
      Further, the voice data is determined as the no-data or the no-sound when an absolute value of an amplitude information of the sound data is equal to or lower than a predetermined value for a predetermined time, and the voice data is determined as the voiced sound when the absolute value of the amplitude information of the sound data is higher than the predetermined value for the predetermined time. Therefore, a processing can be conducted with the small quantity of processing.  
      Further, the voice data is determined as the no-data or the no-sound when an integrated value of square power of the sound data at a predetermined time is equal to or lower than a predetermined value, and the voice data is determined as the voiced sound when the integrated value of the square power of the sound data at the predetermined time is higher than the predetermined value for the predetermined time. Therefore, a precise determining can be performed.  
      Further, the voice data is determined as the no-data or the no-sound when an absolute value of an amplitude information of the sound data is equal to or lower than the first value for a predetermined time, and the voice data is determined as the voiced sound when the absolute value of the amplitude information of the sound data is higher than the second value for the predetermined time. Therefore, the determination of a precise magnitude of the voice data by an average process and the determination of changing from voiced sound to a no-sound can be determined with different threshold values respectively so that a precise determination with the small quantity of processing can be achieved.  
      Further, the voice data is determined as the no-data or the no-sound when an integrated value of square power of the voice data at a predetermined time is equal to or lower than the first value, and the sound data is determined as the voiced sound when the integrated value of the square power of the sound data at the predetermined time is higher than the second value. Therefore, the determination of a precise magnitude of the voice data by an average process and the determination of changing from voiced sound to a no-sound can be determined with different threshold values respectively so that a more precise determination can be achieved.  
      Further, since the second value is set higher than the first value with wide spread, the last data of the voiced sound is prevented from cutting too excess. Also, when the sound data is transit to a voiced sound, the transition of the sound data is pass through an area determined as non-data or no-sound. Therefore, if the second value is rater high, the determination does not become an error.  
      Further, when a predetermined quantity of the voice data is stored in the voice reception buffer, the determining process determines whether the sound data is the no-data or the no-sound or the voiced sound. The discarding process discards the voice data determined as the no-data or the no-sound. Therefore, an inner of the voice reception buffer is arranged when a predetermined quantity of the voice data is stored therein. Therefore, the arranged sound data can be normally transmitted.  
     Second Embodiment  
       FIG. 9  shows a hardware configuration diagram of the camera of the invention.  
       FIG. 10  shows an appearance of the camera of the invention.  
      Numeral  301  designates a camera chip containing the CPU and its peripheral circuits. Numeral  302  designates a flash ROM that stores the program and data for the actions of the camera chip  301 . Numeral  303  designates a working S-DRAM for the cameral chip  301  to act. Numeral  304  designates a CCD and a CMOS chip for converting a taken image into electric signals. Numeral  305  designates an Audio PCM chip for inputting/outputting voice signals. Numeral  306  designates a LANPHY chip for an electric interface at the time of physical connections with a LAN interface. Numeral  307  designates a motor drive chip for moving the camera within a taking range, i.e., a Tilt motor  308  and a Pan motor  309 . There are a microphone for voice inputting and a speaker for voice outputting, although not shown.  
      The camera chip  301  is configured by a CPU  301 - 1 ; a JPEG converter  301 - 2  for converting a taken image in electric signals, into an image of the JPEG format; a G.726 converter  301 - 3  for conversions into the voice data format for the network communication; an MMU (Memory Management Unit)  301 - 4 ; a GPIO (General Purpose Input/Output); and a LAN (Local Area Network)  301 - 6 .  
      This hardware configuration diagram of  FIG. 9  and the camera configuration diagram of  FIG. 1  are mapped, as follows. The camera  10  corresponds to the CCD  304 ; the voice input adjusting circuit  17   a  and the voice output adjusting circuit  18   a  to the Audio PCM  305 ; the portion of the communication control unit  13  to be connected with the LAN to the LANPHY  306 ; the portion for the communication control unit  13  for the control actions to the LAN unit  301 - 6 ; the pan motor  10   b  to the Pan Motor  309 ; the tilt motor  308  to the Tilt Motor  308 ; the image processor  12  to the JPEG converter  301 - 2 ; the voice reception processor  14  and the voice transmission processor  15  to the G.726 converter  301 - 3 ; the camera control unit  10   a  and the control unit  19  for their controls to the CPU  301 - 1 ; and the storage unit  20  to the S-DRAM  303 .  
      Moreover, it is possible to realize: the flash ROM  302  with MX29LV320; the S-DRAM  303  with MT48CM16; the Audio PCM chip  305  with AK2308; the LANPHY chip  306  with ICS1893; the CCD chip  304  with the combination of ICX098, MN5400 and HV7131; and the motor drive chip  307  with LB1937.  
      With these configurations, there can be realized the camera which is enabled to output image information to the network and to output an uninterrupted voice even for a dense communication traffic, by inputting the voice information from the communication terminal and by deciding the magnitude of the voice information received.  
      The invention can be applied to the network system for image transmissions and voice communications by using the voice mapping type network camera.