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    专

    一种基于TaqMan探针法的Beilong病毒检测试剂盒

    KR2006025514U2
    发明人
    张小爱, 刘玮, 陈津津, 丁恒
    受让人
    INST MILITARY MEDICINE ACADEMY MILITARY SCIENCES PLA
    申请人
    Shogo Fujito, Atsushi Shiraishi, Tomohiro Maekawa, Teruyasu Hanagami
    申请号
    400663
    申请日
    2018-11-20
    公开(公告)号
    KR2006025514U2
    公开(公告)日
    2002-03-13
    IPC分类号
    G06T001/00G06T019/00G06F003/01G06F003/0481
    CPC分类号
    -
    优先权号
    199947P
    199960P
    225226P
    275226P
    256399P
    256524P
    258159P
    258511P
    258828P
    259659P
    275226
    275604P
    844861
    优先权日
    2000-04-26
    2000-04-26
    2000-08-13
    2000-08-13
    2000-12-17
    2000-12-17
    2000-12-21
    2000-12-27
    2000-12-27
    2001-01-03
    2001-03-12
    2001-03-12
    2001-04-26
    摘要

    NOVELTY - A protection circuit (ESPR1) provides protection against electrostatic discharge. A diode (D1) is coupled between a supply node (VDDAMIC) and an output terminal (2). A cut-off electronic lock (ITRB) couples the supply node to a supply terminal (3) by reverse biasing the diode to make the voltage of the output terminal float, when a lock-out circuit (VAMIC-AMP) is in an inactive state.

    USE - Electric circuit for electronic apparatus (claimed) e.g. mobile telephone.

    ADVANTAGE - Allows combining protection of output terminal against electrostatic discharges with the possibility to let the output terminal floating, when a lock-out circuit is in an inactive state. Provides protection against electrostatic discharges for the supply terminal through a second discharge device. Ensures proper biasing of transistors.

    DETAILED DESCRIPTION - An INDEPENDENT CLAIM is also included for an electric circuit protecting method.

    DESCRIPTION OF DRAWING(S) - The drawing shows the schematic diagram of electric circuit.

    Output terminal (2)

    Supply terminal (3)

    Diode (D1)

    Protection circuit (ESPR1)

    Cut-off electronic lock (ITRB)

    Lock-out circuit (VAMIC-AMP)

    Supply node (VDDAMIC)

    权利要求
    1. An information block in a multilevel optical data system, said information block comprising: a preamble having (i) a timing acquisition area, (ii) a preamble level calibration area, and (iii) an equalizer training area; a data block; and a postamble.
    2. The information block according to claim 1 , wherein said preamble includes: a time zero locator area; and a block address.
    3. The information block according to claim 1 , wherein said postamble includes timing subunits containing (i) an AGC/timing area and (ii) a DC control area.
    4. The information block according to claim 1 , further comprising: a link-in area prior to said preamble, said link-in area comprising (i) an oscillating mid-level pattern area (ii) and (iii) a link-out area.
    5. An information block in a multilevel optical data system comprising: a preamble having (i) a timing acquisition area, (ii) a preamble level calibration area, and (iii) a preamble equalizer training area; a first data block; a midamble having (i) a midamble level calibration area and (ii) a midamble equalizer training area; a second data block; and a postamble.
    6. The information block according to claim 5 , wherein said preamble includes (i) a time zero locator area and (ii) a block address.
    7. The information block according to claim 5 , wherein said postamble includes timing subunits containing (i) an AGC/timing area and (ii) a DC control area.
    8. The information block according to claim 5 , wherein said first data block and said second data block contain ECC protected data having inner parity and outer parity.
    9. The information block according to claim 5 , further comprising: a link-m area prior to said preamble, said link-in area comprising (i) an oscillating mid-level pattern area and (ii) a link-out area.
    10. A multilevel pattern of marks written to an optical disc, comprising: a preamble having (i) a timing acquisition area, (ii) a preamble level calibration area, and (iii) an equalizer training area; a data block, and a postamble.
    11. The multilevel pattern according to claim 10 , wherein said preamble includes: a time zero locator area; and a block address.
    12. The multilevel pattern according to claim 10 , wherein said postamble includes timing subunits containing an AGC/timing area and a DC control area.
    13. A multilevel pattern of marks written to an optical disc, said multilevel pattern comprising: a preamble comprising (i) a timing acquisition area, (ii) a preamble level calibration area, and (iii) a preamble equalizer training area; a first data block; a midamble including (i) a midamble level calibration area, and (ii) a midamble equalizer training area; a second data block; and a postamble.
    14. The multilevel pattern according to claim 13 , wherein said preamble includes: a time zero locator area; and a block address.
    15. The multilevel pattern according to claim 13 , wherein said postamble includes timing subunits containing (i) an AGC/timing area and (ii) a DC control area.
    16. The multilevel pattern according to claim 13 , wherein said fist data block and said second data block contain ECC protected data having inner parity and outer parity.
    17. The multilevel pattern according to claim 13 , further including: a link-in area prior to said preamble comprising (i) an oscillating mid-level pattern area, and (ii) a link-out area.
    18. The multilevel pattern according to claim 17 , wherein said ECC protected data has punctured inner parity.
    19. The multilevel pattern according to claim 17 , wherein said ECC protected data is convolutionally encoded such that m bits are encoded into n bits.
    20. The multilevel pattern according to claim 17 , where n is greater than m.
    说明书
    [0001]BACKGROUND OF THE INVENTION In the past, telephone connections were generally operated from stationary terminals, which are normally installed in locations which are relatively well shielded from environmental noise (within rooms, telephone booths). However, the explosive growth of mobile telephony has now resulted in a primary factor being the need to set up and maintain telephone connections even in extremely poor acoustic conditions which are varying rapidly and severely. The AF amplifier stages in modern terminals, in particular in mobile radio terminals, are thus equipped with systems, which can be operated manually or under menu control, for adjusting the volume of the receiver capsule. The user can use the appropriate adjustment key or menu function to manually correct the instantaneous receiver volume when the acoustic environmental conditions are varying. These manual corrections are, to a greater or lesser extent, cumbersome, and menu-controlled adjustment processes require the necessity to monitor the display area on the terminal, thus resulting in the necessity to at least briefly interrupt the communication. In addition, the correction process must be repeated whenever there is any significant change in the environmental volume and, in the end, this makes it considerably more difficult to hold a smooth conversation while concentrating on it. Finally, the necessity to manually correct the receiver volume in certain situations, for example while driving a vehicle or while operating a machine, involves a certain safety risk. Thus, in practice, the adjustment capabilities available with the prior art frequently are not used. The present invention is, therefore, directed toward specifying a communications terminal in which the receiver volume is matched easily and more safely to the environmental conditions.
    [0002]SUMMARY OF THE INVENTION The present invention includes knowledge relating to psychoacoustic relationships, which can be outlined as follows: Let us assume there is a voice connection between a subscriber A who is using a mobile telephone in a city area while the environmental volume is fluctuating severely, and a subscriber B has a communication partner, who may himself use any desired terminal. At the subscriber A end, the environmental noise increases (for example, due to an increase in road noise, aircraft noise, a train going past etc.), while the voice connection is in existence. The subscriber A reacts to this by involuntarily increasing the volume of his speech. Conversely, the subscriber A reacts to a reduction in the volume of the environmental noise by involuntarily speaking more quietly. This behavior is supposedly due to the fact that, when two conversation partners located at the same point are talking, the change in the environmental volume affects both of them to the same extent, and the comprehensibility of one's own words for the person one is conversing with, on the one hand, and a volume level which is as reasonable as possible, on the other hand, can be ensured by adaptation of the speech volume. However, in the course of a telephone connection, such behavior which is intended for direct personal conversations makes only limited sense since noise-compensated transmitter capsules in telephones are matched to the nearby area, and changes in the environmental volume are passed on to the subscriber B only in very attenuated form. The subscriber B, therefore, perceives the fact that the subscriber A is raising and lowering his voice as being at least unmotivated, if not annoying. However, the outlined relationship opens up the technical capability to adjust the receiver volume of a telephone by detecting the speech volume of the user on this telephone. Surprisingly, this means that there is no need to separate the environmental noise from the user's speech, which is technically difficult and at least complex. It has been found to be sufficient to detect the speech volume and to actuate the gain adjustment in the AF amplifier stage via a control signal, which is derived from this detection, for adjustment of the gain factor. In one simple and expedient embodiment, the transmitter capsule of the telephone is directly followed by a level or voltage measurement device, which detects the speech volume. Adapted processing of the primary measurement signal in order to avoid undesirable gain changes and control oscillations in the speech/listening system resulting from natural volume fluctuations while the user is speaking and during pauses in speech require a certain amount of care. For this purpose, the volume signal must expediently first of all be averaged using a suitably selected time constant, for example in the region of seconds. Pauses in speech must be considered separately, for example by deactivation of the signal evaluation when a predetermined threshold level is undershot. In general, the established mobile radio standards provide functions which can be used for this purpose. For example, the functionality of a GSM voice coder and decoder (CODEC) include what is referred to as the DTX Mode (Discontinuous Transmission), in which the transmitter is activated only when a specific speech frame (20 ms for GMS) actually includes spoken information. In principal, this function also can be used to exclude pauses in the user's speech from the evaluation of the speech volume. In an embodiment which can be flexibly matched to the normal speech characteristics of the user or to the area in which the telephone is being operated at that time, the averaging interval and/or the deactivation threshold value are/is adjustable manually or under menu control. Thus, for example, the deactivation threshold value can be increased when using a mobile station with a sensitive transmitter capsule when the environmental volume level is high (in traffic concentrations, in production areas where the noise levels are high, in coastal areas etc.), to take account of this, and the individual adjustment of the averaging interval allows the preselection of a fairly sensitive or inert response for the gain setting, as a function of the user's personal lifestyle. In a further advantageous embodiment, parts for detecting the speech volume or for gain adjustment are designed such that they can be switched off as a function thereof, so that the telephone also can be operated with a fixed AF gain setting, or can be changed to manual gain adjustment. In a further embodiment, the area in which the volume level of the receiver capsule can vary as a function of the environmental noise can be predetermined as a subregion (“window”) within the overall range of the feasible receiver capsule volume, or the overall AF level range. In addition, in this embodiment, the presence of parts for shifting this window, as required, within the overall level range is advantageous. The parts for carrying out this shift can once again be operated manually by a switch, or under menu control. As an alternative to this, automatic shifting is feasible, as a function of a defined parameter, for example the reception field strength in a mobile station. Detection of the dynamic response in the user's speech can also, in one specific embodiment, provide an additional control signal not only for positioning the “level window” but also for specific adaptive averaging of the speech volume. One advantageous development of the present invention also provides for the effect of varying the speech volume when the environmental noise level changes, which is irritating to the conversation partner in a telephone connection (and which the other subscriber receives only to a very restricted extent especially when using a noise-compensated microphone in the transmitter capsule of his conversation partner) to be at least partially compensated for. For this purpose, a control stage is expediently provided which is connected to the input of the AF amplifier stage, where the undesirably fluctuating signal level is present. In one preferred embodiment, this control stage also can be selectively activated or deactivated via a switching element or under menu control, in order to allow matching to the user's listening characteristics. Additional features and advantages of the present invention are described in, and will be apparent from, the following Detailed Description of the Invention and the Figures.
    [0003]BRIEF DESCRIPTION OF THE FIGURES FIG. 1 shows in block diagram form various components of a mobile telephone to which the present invention is directed.
    [0004]DETAILED DESCRIPTION OF THE INVENTION FIG. 1 shows, in the highly simplified form of a functional block diagram, those components which are essential to the explanation of the present invention in a mobile telephone 1 , including a receiver capsule 3 with a load speaker 3 a , a transmitter capsule 5 with a noise-compensated microphone 5 a , and an AF amplifier stage 7 with an AF signal input 7 a. In addition to being connected to the AF amplifier stage 7 , the AF signal input 7 a is connected via a first electronically controllable switch 9 to an AF level control stage 11 , whose output is connected to a first control input 7 b of the AF amplifier stage 7 . The transmitter capsule 5 is connected via a second electronically controllable switch 13 to a voltage measuring device 15 for detecting the actual level of the speech volume at the microphone 5 a . The output of the voltage measuring device 15 is connected to an averaging stage 17 , which is connected via a control input to the output of an averaging interval adjustment device 19 . On the output side, the averaging stage 17 is connected firstly to a gain adjustment stage 21 , and secondly to one input of a speech level component unit 23 , whose other input is connected to a speech level threshold value memory 25 , and whose output is connected via a third electronically controllable switch 27 , for control signal purposes, to a fourth switch 29 , which makes or breaks the connection between the averaging stage 17 and the gain adjustment stage 21 as a function of the output signal from the speech level comparator unit 23 and the position of the third switch 27 . The gain adjustment stage 21 is associated, via a control input, with a gain range preselection stage 31 , and the gain selection stage is connected on the output side to a second control input 7 c of the AF amplifier stage 7 . The output of the AF amplifier stage 7 is connected to the loudspeaker 3 a of the receiving capsule 3 . The first, second and fourth controllable switches 9 , 13 and 29 as well as the averaging interval adjustment device 19 and the gain range preselection stage 31 are actuated via a microcontroller 33 of the mobile telephone, which is connected in the normal way to its input keypad or keyboard 35 and to the display 37 , and which has an associated program memory area (ROM) 33 a and a main memory area (RAM) 33 b. This arrangement operates as follows: the AF level control stage 11 , which fundamentally operates in the same way as automatic level control, as is known per se, produces (depending on the specific configuration) partial or complete compensation for fluctuations in the input level at the AF signal input 7 a , provided the first switch 9 is closed, and this function is hence activated. This compensates, in particular, for increases or reductions in the speech volume of the conversation partner in a telephone connection resulting from environmental noise. If the second switch 13 is closed, and the function of automatic matching of the receiver volume to the speech volume is thus activated, the voltage measuring device 15 detects the speech volume level, with the output signal from the voltage measuring device being subjected in the averaging stage 17 to averaging using an averaging time interval which is preselected via the averaging interval adjustment device 19 and is preferably in the range between 1 and 10s. This prevents the AF gain from being influenced by brief pauses in speech between individual words or sentences or by natural level fluctuations in the course of the conversation, with the preselection of an averaging time interval via the keyboard or keypad 35 , the microcontroller 33 and the averaging interval adjustment device 19 allowing specific matching to normal speech characteristics of the user or the specific situation in which the mobile telephone is being operated. The output signal from the averaging stage 17 is subjected, in the speech level comparator unit 23 , to level discrimination using a threshold value that is stored in the speech level threshold value memory, provided this function is activated by closing the third switch 27 . If the volume level of the microphone 5 a falls below the predetermined threshold value, then opening the fourth switch 29 results in the corresponding conversation phase being excluded from the automatic gain adjustment process. As soon as the speech level is once again above the threshold value, the output signal from the comparator unit 23 changes, following which the fourth switch 29 is closed again, and the subsequent phase of the conversation is included in the process by transmitting the output signals from the averaging stage 17 to the gain adjustment stage 21 for the AF gain adjustment. In addition, as for the other functions and possibly including menu control with the aid of the display 37 , the gain range preselection stage 31 also results in preselection, via the keyboard or keypad 35 and the microcontroller 33 , of a gain range or “level window” within whose boundaries the gain adjustment stage 21 is intended, in response to the output signal from the averaging stage 17 , to adjust the gain of the AF amplifier stage 7 via its second control input 7 c. The embodiment of the present invention is not restricted to the described exemplary embodiment, but is also feasible in a large number of modified versions. For example, in simplified embodiments, there is no need for some or else all of the described switching and adjustment functions, or these functions can be provided by conventional electrical switches or adjustment controllers. The present invention can be implemented not only in a mobile telephone but also, in particular, in a cordless telephone or in a normal land line terminal. It is self-evident that the described functions can to some extent be implemented in the form of software. In this context, the wide range of speech processing functions in the protocols of modern mobile radio systems offer worthwhile capabilities which largely results in there being no need for any additional hardware complexity. If a digital signal processor (DSP) is available, the functions described above can be carried out with high precision, and dependent only on the software. The method of the present invention can be applied, in addition to telephones, in general to all communications appliances using simplex or duplex speech operation (that is to says also including, for example, door interphones or the like). Although the present invention has been described with reference to specific embodiments, those of skill in the art will recognize that changes may be made thereto without departing from the spirit and scope of the invention as set forth in the hereafter appended claims.
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