• 专利附录
  • 专利说明
  • 权利要求
  • 类似技术
  • 同族专利
  • 引用文献
  • 法律状态
  • 优先权
    • 专利摘要:
    1. A digital conferencing method that includes sampling the subscriber speech signals and non-linear coding of these discretized speech signals, converting non-linearly-encoded speech signals into linear-coded speech, adding the linear-coded speech signals of different subscribers, and converting it into non-linearly coded speech signals signals, characterized in that, in order to reduce distortion and increase the intelligibility of speech, discretization. The real speech signals of different subscribers are integrated and compared with each other, while before adding, they are. line-coded speech signals of different subscribers, attenuate the specified signals of all subscribers, except the signal of the subscriber with the highest intensity. (L with
    公开号:SU1210677A3
    申请号:SU823522504
    申请日:1982-12-09
    公开日:1986-02-07
    发明作者:Эшманн Йоахим；Цанциг Юрген
    申请人:Интернэшнл Стандарт Электрик Корпорейшн (Фирма)；
    IPC主号:
    专利说明:

    2. A device for digital configuration, containing in series the first code converter, the adder, the second code converter and the parallel code to serial converter, the first memory block, whose input is connected to the output of the adder, and the output to the additional input of the adder, and serial to parallel converter, characterized in that a first comparator, a counter, a second comparator and a damping control unit, the output of which is connected in series, are entered in it; inen with the control input of the first code generator, as well as the second block
    The invention relates to communication technology and can be used in digital telephone switchboard installations with pulse code modulation (CMM).
    The purpose of the invention is to reduce distortion and increase speech intelligibility.
    FIG. Figure 1 shows the electrical circuit diagram of a device for implementing a digital conference method; in fig. 2 is a device that generates control signals that are fed to separate units to control the temporal sequence of processing processes.
    The digital conferencing method is as follows.
    The speech signals of individual conferencing subscribers are sampled, integrated over time, then compared to each other subjected to nonlinear coding and conversion of nonlinear-coded to linear-coded, followed by linear-coded speech signals of all subscribers, except for the signal of the subscriber with the highest intensity are weakened. Then, the linear-coded speech signals of different users are added, the input of which is connected to the input of the first comparator and the output of the serial-to-parallel converter, and the output of the second memory block is connected to the information input of the first code converter.
    3. The device according to claim 2, wherein between the output of the serial to parallel converter and the input of the second memory block is included a free channel suppression unit, the auxiliary input of which is connected to the additional output of the serial to parallel converter.
    Thus, each subscriber receives the sum of the voice signals of other subscribers, after which the transformation of linearly coded speech signals into non-linearly coded signals is performed.
    If n participants participate in the conference, n different additions are made in the device for implementing the proposed method, with n-1 voice signals being added each time. The proposed device is able to form six independent circularly included groups, each of which contains up to five subscribers. The speech signals of various subscribers are estimated during the integration time, and the speech signal of that subscriber, which is defined as the loudest voice subscriber, is summed without fading, while the voice signals of all other subscribers are summed after fading by 12 dB .
    A device for implementing a digital conference method comprises a serial code to parallel converter 1, first 2 and second 3 code converters, adder A, converter 5
    parallel code in the sequence 1, the first 6 and second 7 memory blocks, the counter 8, the first 9 and second 10 comparators, the attenuation control unit 11, the free channel suppression unit 12.
    The device works as follows.
    In switchboard time-division CMM installations, the incoming bit sequence is subdivided into 32 channels or time gaps per frame, each frame having 256 bits. Each speech test, i.e. Each voice reading value is encoded in one CMM word containing 8 bits, so that 256 quantization steps are obtained. The read frequency is 8 kHz. Of the 32 channels, 30 channels are working channels, while Channel No. O is for frame synchronization, and Channel No. 16 is for signaling.
    The sequence of bits enters the input of the serial code converter 1 into a parallel one; the last eight-bit parallel code from the output of the last code arrives at the input of the free channel suppression unit 12. The additional input of block 12 for the suppression of free channels from a serial to parallel converter 1 receives two protocol bits representing the format of each channel. Free channel suppression unit 12 establishes empty channels by controlling bit formats and suppresses them for further processing. From the output of block 12 for suppressing free channels, the words consisting of 8 bits are fed to the input of the second memory block 7, in which they are intermediately stored, and to the input of the first comparator, where are compared in groups, i.e. all voice signals of one circular communication are compared. For each channel and thus for each circularly connected subscriber, a counter 8 is provided. After each comparison of the speech signals of one circularly included subscriber group, the readings of counter 8, provided for the subscriber with the loudest voice, increase by one unit, while indications provided for others

    the subscribers of the counters are reduced by one unit. Counters 8 count from zero to 63 units. Thus, a maximum of 32 frames will pass, i.e. 32x12 ms, until the counter of the subscriber with the loudest voice overtakes the counter of 8 other subscriber whose voice was earlier than the loudest.
    In order to determine the subscriber, the speech signal of which has the highest intensity, i.e. the subscriber with the loudest voice, the readings of the counter 8 of the circularly connected subscribers are compared in groups with each other in the second comparator 10. On the basis of this comparison, the second comparator 10 delivers control bits, the so-called damped bits, to the input of the attenuation control unit 11, in which -, rum, they are intermediately memorized. From the last, the damped bits are fed to the control input of the first, a 2 code converter, which performs two functions. The first function is that it converts non-linearly encoded speech signals obtained by encoding analog speech signals into linearly-encoded speech signals necessary for addition without distortion. Nonlinear coded words CIM consist of 8 bits, while linearly coded speech signals contain 13 bits each. From the output of the second memory block 7, the compressed speech signals are fed to the input of the first 2 code converter, from the output of which voice signals are already in linear-coded form.
    The second function is that in the first 2 code converter all voice signals are attenuated except for the voice signal of the subscriber with the loudest voice. The first converter of 2 codes is designed as a programmable constant-value accumulator, in the proposed device, attenuation of 12 dB is chosen, but another one can be chosen.
    Attenuation control unit 11 determines which speech signal to attenuate and which not. The accumulation of the results of comparisons made in the first comparator 9 in the counters 8 provided for individual subscribers and the execution of these counters gives the first integration of the estimated speech signals over time. The first generator of 2 codes to the adder 4 receives the consecutive speech signals and the decaying bits belonging to them of the circularly connected subscribers for summing into the sum of the matrix 4. After each separate addition, the sum signals are intermediate stored in the first memory block.
    After the addition, the speech signals are sent to the second converter of the 3 codes, where they are converted into nonlinearly coded speech signals, which are then fed to the input of the converter 5 of the parallel code to the serial one. From the output of the last word, the CMMs arrive at the output of the circular circuit in a sequential form, namely, through the appropriate switching unit to the communication network and from the last to the telephone subscribers.
    There is a shift of six channels J in time, i.e., between the channels of the CMM entering the circular scheme and the channels leading from it. shift about 24 µs.
    The proposed circular circuit can also be connected. In this case, for example, in a subscriber unit with additional devices, either the caller is connected to the busy subscriber input of the called subscriber, or the attendant is switched on to notify about the connection from the public telephone network to the existing telephone connection.
    ten
    15
    H
    Due to the fact that in the proposed circular scheme all speech samples, except for one, are subjected to damping before summing, the considerable overflow size of the amplitude range allowed for the total speech signal is reduced, as well as the likelihood of such overflow.
    MHtSMz
    FIG. g
    U / 7 / i
    权利要求:
    Claims (3)
    [1]
    1. A method of digital conferencing, including the sampling of speech signals of subscribers and non-linear coding of these sampled speech signals, the conversion of non-linearly encoded speech signals to linearly encoded, the addition of linearly encoded speech signals of various subscribers and the conversion to non-linearly encoded speech signals, characterized the fact that, in order to reduce distortion and improve speech intelligibility, is discretized. Different voice signals of various subscribers integrate and compare with each other, while before adding linearly encoded speech signals of various subscribers attenuate the indicated signals of all subscribers except the signal of the subscriber with the highest intensity.
    I
    FIG. one
    [2]
    2. A device for digital conferencing, containing a series-connected first code converter, an adder, a second code converter and a parallel code to serial converter, a first memory unit whose input is connected to the output of the adder, and the output - with an additional input of the adder, and the Serial code converter in parallel, characterized in that the first comparator, counter, second comparator and attenuation control unit, the output of which is connected the control input of the first preobazovatelya codes, and the second unit / memory input is connected to the input of the first comparator and the output of the serial to parallel converter and the second output of the block memory is connected to the data input of the first code converter.
    [3]
    3. The device according to claim 2, characterized in that between the output of the serial code converter in parallel and the input of the second memory block, a free channel suppression unit is included, the additional input of which is connected to the additional output of the serial code to parallel converter.
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    同族专利:
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    ES518084A0|1983-09-01|
    JPS58150357A|1983-09-07|
    EP0081799B1|1988-05-04|
    NZ202626A|1985-07-31|
    AU9113182A|1983-06-16|
    NO824084L|1983-06-13|
    IN157468B|1986-04-05|
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    BR8207094A|1983-10-11|
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    NO157839C|1988-05-25|
    IT1154391B|1987-01-21|
    IT8224552D0|1982-12-02|
    PH20311A|1986-11-25|
    RO84971A|1986-07-30|
    AU553691B2|1986-07-24|
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    ES8308667A1|1983-09-01|
    CA1206237A|1986-06-17|
    US4488291A|1984-12-11|
    HK67990A|1990-09-07|
    KR880002168B1|1988-10-17|
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    EP0081799A1|1983-06-22|
    TR21922A|1985-11-07|
    YU272882A|1985-10-31|
    ZA828846B|1983-09-28|
    FI824200A0|1982-12-07|
    YU43810B|1989-12-31|
    JPH0129345B2|1989-06-09|
    AT34058T|1988-05-15|
    SG2889G|1989-11-17|
    MX151911A|1985-05-02|
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    引用文献:
    公开号 | 申请日 | 公开日 | 申请人 | 专利标题
    GB1301905A|1970-06-04|1973-01-04|
    US3883697A|1973-10-18|1975-05-13|Itt|Digital conference circuit|
    US3937898A|1974-07-18|1976-02-10|International Telephone And Telegraph Corporation|Digital conference bridge|
    DE2445094C3|1974-09-20|1978-11-02|Siemens Ag, 1000 Berlin Und 8000 Muenchen|Method for establishing conference calls in a PCM time division multiplex telephone exchange|
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    US4153817A|1977-11-25|1979-05-08|Stromberg-Carlson Corporation|Digital conference circuit|
    US4190742A|1978-06-05|1980-02-26|Siemens Aktiengesellschaft|Process and apparatus for producing conference connections in a PCM time multiplex switching system|
    US4267593A|1979-06-15|1981-05-12|Wescom Switching, Inc.|Method and means for digital conferencing|
    JPS56157164A|1980-05-07|1981-12-04|Nec Corp|Conference system|
    US4387457A|1981-06-12|1983-06-07|Northern Telecom Limited|Digital conference circuit and method|DE3329779C2|1983-08-18|1985-06-27|Philips Patentverwaltung Gmbh, 2000 Hamburg|Method and circuit arrangement for establishing conference connections in a switching system|
    JPS60148263A|1984-01-12|1985-08-05|Nippon Telegr & Teleph Corp <Ntt>|Conference telephone system|
    JPH034148B2|1984-03-29|1991-01-22|Nippon Denshin Denwa Kk|
    DE3427303C2|1984-07-20|1988-12-22|Siemens Ag, 1000 Berlin Und 8000 Muenchen, De|
    US4606021A|1984-08-17|1986-08-12|Itt Corporation|Digital conference circuit|
    JPH0237152B2|1984-10-11|1990-08-22|Iwatsu Electric Co Ltd|
    US4658398A|1984-10-29|1987-04-14|Gte Laboratories Incorporated|Framed digital voice summing for teleconferencing|
    US4648089A|1984-10-29|1987-03-03|Gte Laboratories, Inc.|Digital voice summing for teleconferencing|
    US4686512A|1985-03-01|1987-08-11|Kabushiki Kaisha Toshiba|Integrated digital circuit for processing speech signal|
    US4716585A|1985-04-05|1987-12-29|Datapoint Corporation|Gain switched audio conferencing network|
    WO1996004744A1|1994-08-04|1996-02-15|National Semiconductor Corporation|System and method for conducting conference calls in a communication system utilizing voice compression|
    US5483588A|1994-12-23|1996-01-09|Latitute Communications|Voice processing interface for a teleconference system|
    JP2007096555A|2005-09-28|2007-04-12|Nec Corp|Voice conference system, terminal, talker priority level control method used therefor, and program thereof|
    US20070266092A1|2006-05-10|2007-11-15|Schweitzer Edmund O Iii|Conferencing system with automatic identification of speaker|
    CN102598644B|2009-11-06|2015-05-27|飞思卡尔半导体公司|Conference call system and method|
    法律状态:
    优先权:
    申请号 | 申请日 | 专利标题
    DE3148886A|DE3148886C1|1981-12-10|1981-12-10|Method and circuit arrangement for establishing a conference connection|
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