JPH03229530A - Method and system for time-division communication of moving body communication - Google Patents

Abstract

PURPOSE:To reduce transmission output and to effectively use a frequency by taking the number of multiplex for TCM(time division compression multiplex) signal, the time length of one frame and a highest frequency which an original signal has as parameters, and clearly introducing a multiplex load gain through the use of a sampling theorem. CONSTITUTION:The multiplex load gain which the TCM(time division compression multiplex) signal transmitted from a radio base station 30 has is caused to correspond to a multiplex load gain which an FDM(frequency time division multiplex) signal has. Thus, respective sound signals flowing in respective channels CH1, CH2,..., CHn of FDM are expressed in the form of a function. Then, the gain of the multiplex signal is introduced by using the sampling theorem. Thus, a system can be designed so that the multiplex load gain which is substantially the same as that in a case when transmission is executed from the radio base station 30 can be obtained even in a case when a moving radio unit 100 executes transmission and the radio base station 30 executes reception.

Description

[Detailed description of the invention]

[Industrial Application Fields] +Y, 'A, Light are multiple loads of time-compressed multiplex signals, which are modulation signals, in time-division communication methods and systems for wireless communication channels suitable for digital communication networks (ISDN) in mobile communications. Regarding effective use of profits. More specifically, given a certain radio channel, one of the many mobile radios that are compatible with the digital communication network within the service area establishes a radio link with an opposing radio base station. If another mobile radio starts communicating with another radio base station using the same radio channel while communicating, the communication A method for eliminating adverse effects on communication between mobile radio equipment and radio base stations, and at the same time improving the effective use of frequencies by reducing transmission output, and an economical system using the method. Provide/
ν. [Prior Art] In mobile communication using voice using a small zone method, a method employing a time division time compression multiplex signal is described in the following document. Literature 1. Ito ``Study of mobile phone system - Time division time compression F
Proposal of M modulation method-” IEICE technical report RC589-11
July 1989 document 2. Ito “Study of mobile phone system - Time division time compression F
Theoretical study of M modulation system” IEICE technical report RC389-39
In other words, in Document 1, a transmission signal (baseband signal) is divided into predetermined time intervals and stored in a memory circuit, and when read out, the speed is n times faster than the speed at which it is stored in the memory circuit. It is built into the mobile radio R and the radio base station in order to read data in a predetermined time slot at high speed, angle-modulate or amplitude-modulate the carrier wave with the signal accommodated in the time slot, and transmit and receive the signal intermittently in time. a wireless receiving circuit having a receiving mixer that communicates with each other, a wireless transmitting circuit having a transmitting mixer, a synthesizer applying voltage to the receiving mixer of the wireless receiving circuit, and a synthesizer applying voltage to the transmitting mixer of the wireless transmitting circuit. On the other hand, a switch circuit is provided to intermittent the output of each applied synthesizer, and to synchronize this intermittent state for both transmission and reception, and also to synchronize the same intermittent transmission and reception as above with that of the mobile radio device for the radio base station that communicates oppositely. In order to extract only the signals accommodated in the predetermined time slots, the receiving side opens and closes the radio receiving circuit to receive the signals, demodulates the signals, stores the obtained signals in the storage circuit, and An example of a system has been reported in which a system is constructed in which it is possible to reproduce the baseband signal, which is the original signal that has been transmitted, by reading out the signal at a low speed that is 1/n of the speed at which it is stored in this memory circuit. There is. In addition, in Document 2, there is a study of adjacent channel interference and co-channel interference, which are problems in the above-mentioned TCM (time division time compression multiplexing)-FM method applied to a small zone. Possibilities of system realization are shown by appropriate selection of system parameters. Furthermore, the multiple load gain of a so-called frequency division multiplexed signal, which is obtained by converting the frequency of an audio signal and multiplexing it so that it does not overlap on the frequency axis, is described in, for example, the following documents 3 and 4.
is shown. Literature 3. B, D, Ho1brook, J, T, Di
xon: LoadRating Dingheory
for ) lultichannel Ampli
fiersBSTJ, 18. Oct., 1939 Reference 4. C.B., Feldman et al.”Band Widt
H and D transmission performance
mance” BS Ding J, July 19
Pages 49490 to 595, Figure 8 was created from Figures 7 of the above-mentioned document 3, and Figure 9 was quoted from page 495 of the above-mentioned document 4. This shows that it is possible to obtain substantially the same multiple load gain as . The reason why multiple load gain can be obtained and its application to wireless angle modulation will be briefly explained below. The level of an active telephone line carrying telephone signals varies from person to person, gender, and subscriber line length; There is always a gap. Also, while the other party is talking, the other party does not speak and no signal is added to one direction. Do not speak during exchange connection. Therefore, the individual signal levels must be diverse, and a composite signal of these cannot be easily obtained. However, clarifying this is the most important and fundamental problem in creating a relay line that maintains distortion, crosstalk, quasi-crosstalk, noise, etc. to satisfactory values. Therefore, it has been studied by many people. FDM method (SS: 5 in Ole 5
The level of the system (method applying ideband) is such a synthesis of voices, and the probability that each voice overlaps at the same time is rare, and while the number of communication channels N is small, each voice fluctuates greatly, but the influence on the synthesized signal is not direct. As the number of multiplexes increases, the individual effects become less direct and average out stochastically. Therefore, the peak value of the composite signal increases very slowly as the number of communication paths increases. This is B, D, H. Brook and J. T. Dixon (reference 3, supra) statistically determined this for telephone calls in the United States. According to the results, the change in power of the limited wave having the same peak value as the peak value of the multiplexed signal is as shown in FIG. To show how small the increase in the peak value of a multiple telephone signal is, when compared with the sum of the peak voltages of the individual signals, it becomes as shown in the "multiple load gain" in Figure 8. That is, for example, in the 960 telephone line system, is the same peak voltage as when the 6 channels are simultaneously loaded to the maximum and the signal of 954 channels is not loaded.In the SS-FM system, the voltage fluctuation of the composite signal becomes a frequency deviation, so the composite signal When the peak frequency deviation of is set to a certain value, as the number of multiplexed channels N increases, the modulation for each channel is increased by the multiple load gain shown in Figure 8, compared to when the voltages of each speech signal are summed. The index can be made large, and S given when the leading frequency deviation is set to an arbitrary value, /
It is improved by that much more than N. [Problem to be solved by the invention] In the system construction examples shown in the above-mentioned documents 1 and 2,
There is no disclosure of the existence of the multiple load gain of time-division time compression multiplexed signals transmitted from a radio base station to a large number of mobile radios, and this multiple load gain is not utilized. Furthermore, no consideration is given to suitability for digital communication networks. Therefore, if this multiload gain analysis had been performed, many advantages would be gained in system design, such as the reduction in transmit power level possible by increasing the depth of frequency modulation. , Ding C
Specifically realize advantages such as ease of designing an amplifier for amplifying the M signal, realization of an economical amplifier by expanding the operating level setting range, and economy by relaxing the rating conditions of mixers, resistors, and capacitors. There was an issue that needed to be solved that could not be done. What is disclosed in Documents 3 and 4 is a so-called frequency division multiplex signal in which audio signals are frequency-converted and multiplexed so that they do not overlap on the frequency axis.
However, it is not applicable to time-division time compression multiplexing (DCM) signals, and the existence of multiple load gain is also unknown. There was a problem to be solved in that many advantages (the same as in the cases of Documents 1 and 2) that would have been obtained in the system design cannot be concretely realized if the system were designed. [Means for solving the problem] The multiplexing number (number of communication paths) of TCM (time division time compression multiplexing) signals, the time length of one frame, and the highest frequency of the original signal are taken as parameters, and the multiple load gain of the TCM signal is calculated. Using the sampling theorem, we clearly derived the relationship with the multiple load gain in FDM (frequency division multiplexed signal), and made it possible to put it into practical use in digital communication networks. [Effect 1] Since it has become clear that multiple load gain exists even in CM signals, it is now possible to specifically calculate multiple load gain using various design parameters of systems including digital communication networks. By increasing the degree of modulation of FM (PM) modulation, it is possible to gradually reduce the transmission output while keeping interference within the permissible value. Therefore, the design of the amplifier becomes easy, and the rated values of passive circuits such as mixers, resistors, and capacitors can be lowered, making it possible to construct an economical system. [Embodiment] FIG. 1A, FIG. 1B-1, and FIG. 10-1 show a system configuration for explaining an embodiment of the present invention. In FIG. 1A, 10 is a general telephone network, and 20 is a gateway exchange for connecting the telephone network 10 and a wireless system. 30 is a wireless base station and a barrier switch 20
, a circuit for converting signal speeds, a circuit for assigning and selecting time throwers, a control section, etc., and a control section for setting and canceling the wireless line. ~100-n> and has a wireless transmitting/receiving circuit that sends and receives wireless signals g.Here, between the gateway exchange 20 and the wireless base station 30,
There are transmission lines for transmitting communication signals 22-1 to 22n including communication signals of communication channels CF11 to CHn and control signals. FIG. 1B-1 shows a circuit configuration of a mobile radio device 100 that communicates with a radio base station 30. Received signals such as control signals and call signals received by the antenna section enter a radio receiving circuit 135 that includes a receiving mixer 136 and a receiving section 137, and the communication signal that is the output thereof is sent to a speed restoration circuit 138.
This is input to the control section 140 and tarok regenerator 141. The tarlock regenerator 141 regenerates a clock from the received signal and applies it to the speed recovery circuit 138, the control section 140, and the timing generator 42. The speed restoration circuit 138 restores the speed (pitch in the case of an analog signal) of the compressed and segmented communication signal in the received signal and inputs the restored signal to the telephone section 101 and the control section 140 as a continuous signal. The communication signal output from the telephone unit 101 is divided into predetermined time intervals by a speed conversion circuit 131, and the speed (pitch in the case of an analog signal) is made high (compressed) and sent to a transmission mixer 133. The signal is applied to the wireless transmission circuit 132 including the section 134. The output of the modulator included in the transmitting section 134 is transmitted to the transmitting mixer 13
3, the signal is converted to a predetermined radio frequency, transmitted from the antenna section, and received by the radio base station 30. In order for the mobile radio R100 to send a radio signal to the radio base station 30 using a time slot that is permitted to be used, timing information from the timing generator 142 shown in FIG. It is necessary that you have obtained the necessary information. In this timing generator 142, the clock regenerator 14
1 and a control signal from the control section 140, the transmission/reception intermittent controller 123. It supplies necessary timing to the speed conversion circuit 131 and speed restoration circuit 138. This mobile radio device 100 further includes a synthesizer 121.
-1 and 121-2, and selector switches 122-1, 1
22-2, and a transmission/reception intermittent controller 1 that generates signals for switching the changeover switches 1221 and 122-2, respectively.
23 and a timing generator 142 , and the synthesizers 121 - 1 and 121 - 2 , the transmission/reception intermittent controller 123 , and the timing generator 142 are controlled by the control section 140 . Each synthesizer 121-1.12
1-2 is supplied with a reference frequency from a reference crystal oscillator 120. A wireless base station 30 is shown in FIG. 1C-1. N-channel communication signals 22-1 to 22-2 with the gateway exchange 20
2-n is a signal processing unit 31 that serves as an interface on a transmission path.
connected to. Now, the communication signal 22- sent from the barrier switch 20
1 to 22-n are input to the signal processing unit 31 of the wireless base station 30. The signal processing section 31 is equipped with an amplifier for compensating for transmission loss, and also performs so-called 2-wire to 4-wire conversion. That is, the input signal and the output signal are mixed and separated, and the input signal from the barrier switch 20 is sent to the signal speed conversion circuit group 51. Also, signal speed restoration circuit group 3
The output signal from 8 is transmitted to the barrier exchange 20 by the signal processing unit 31 using the same transmission path as the input signal. Among the above input signals from the barrier switch 20, the signal speed conversion circuit group 5 including many signal speed conversion circuits 51-1 to 51-n
1 and undergoes speed (pitch) conversion at predetermined time intervals. Furthermore, the signal transmitted from the wireless base station 30 to the barrier exchange 120 is determined by the output of the wireless receiving circuit 35 being transmitted to the signal selection circuit group 39.
is input to the signal speed restoration circuit group 3B via the signal speed (
pitch) is converted and input to the signal processing section 31. Now, the control of the radio reception circuit 35 or the output of the call signal is performed by a signal selection circuit 39 that selects a signal for each time slot.
-1 to 39-n is input to the signal selection circuit group 39,
Here, speech signals are separated corresponding to each speech channel CH1 to CHn. This output undergoes signal speed (pitch) restoration in a signal speed restoration circuit group 38 including signal speed restoration circuits 38-1 to 38-n provided for each channel, and then is input to the signal processing section 31. , after undergoing 4-wire to 2-wire conversion, this output is sent to the barrier switch 20 as communication signals 22-1 to 22-2.
2-n. A detailed circuit configuration example of the signal processing section 31 is shown in FIG. 1D-1. Here, 87-1 to 87-n are 4 wires-2
Line converters, 88-1 to 88-n are A/D converters, 89-
1 to 89-n are D/A converters. A/D converter 88
-1 to 88-n are each signal speed restoration circuit 38-1
The analog signals from ~38-n are converted into digital signals and sent through the 4-wire and 2-wire converters 87-1 to 87-n as communication signals 2.
The digital communication signals 22-1 to 22-n are sent to the barrier exchange 20 as 2-1 to 22-n, and are sent from the barrier exchange 120 as digital communication signals 22-1 to 22-n.
n is applied to the D/A converters 891 to 89-n via the 4-wire to 2-wire converters 87-1 to 87-n, respectively, and converted into analog signals, and then sent to the signal speed conversion circuits 51-1 to 51-n. 5
1-n. Next, the functions of the signal speed conversion circuit group 51 will be explained. By storing input signals such as audio signals and control signals divided into a certain length of time in a storage circuit, and changing the speed when reading them out, for example, reading them out at a speed 15 times faster than when they were stored, the signal can be read out. It becomes possible to compress the time length. The principle of the signal speed conversion circuit group 51 is the same as when playing back audio recorded by a tape recorder at high speed, and in reality, for example, a CCD (Char
geCoupled Device), BBD
(Bucket BrigadeDeViCe > can be used, and the memory used in television receivers and tape recorders that compress or expand the time axis of conversations can be used (References: Koita et al. Tape recorder pad that compresses/expands the shaft
Nikkei Electronics July 26, 1976 92-
133 pages). The circuit using the COD/BBD exemplified in the signal speed conversion circuit group 51 can be used as it is in the signal speed restoration circuit group 38 as described in the above-mentioned literature. This can be achieved by making the reading speed slower than the writing speed by receiving a timing signal from a timing generator 42 that generates timing based on the clock and a control signal from the control unit 40. The control or audio signals outputted from the barrier switch 20 via the signal processing unit 31 are input to the signal speed conversion circuit group 51, and after speed (pitch) conversion processing is performed, signals are assigned to each time slot. It is applied to the signal allocation circuit group 52. The signal allocation circuit group 52 is a buffer memory circuit that stores one section of high-speed signals output from the signal speed conversion circuit group 51, and receives timing information from the timing generation circuit 42 given by instructions from the control section 40. Then, the signal in the buffer memory is read out and transmitted to the wireless transmission circuit 32. As a result, when viewed in terms of channel correspondence, communication signals are arranged in series without overlapping in chronological order, and when all control signals or communication signals, which will be described later, are implemented, they appear as if they were continuous signal waves. become. The state of this compressed signal is shown and explained in FIGS. 2A and 2B. The output signal of the signal speed conversion circuit group 51 is sent to the signal assignment circuit group 5.
2 and are given time slots in a predetermined order. SDl in Figure 2A(a), 50
2- and SDn indicate that the speed-converted communication signals are allocated to each time slot. Here, a synchronization signal and a control signal or a communication signal are accommodated in one time thrower l~ as shown in the figure. If the call signal is not implemented, only the synchronization signal is added and the empty slot signal is added to the call signal portion. In this way, as shown in FIG. 2A(a), the wireless transmitting circuit 3
2, signals forming one frame are applied to the modulation circuit in time slots SD1 to SDn. The time-series multiplexed signal to be transmitted is angularly modulated in the radio transmission circuit 32, and then sent out into space from the antenna section. When making or receiving a telephone call, the wireless base station 30
Regarding the transmission of control signals between the mobile radio device 100 and the mobile radio device 100, it is possible to use either within the telephone signal band or outside the telephone signal band. Figure 3A shows these frequency relationships. In other words, in the figure (a), it is an example of an out-of-band signal, and as shown in the figure,
The low frequency side (250H2>) or the high frequency side (3850Hz>) can be used. This signal is used, for example, when you want to send a control signal during a call. In Fig. 3A (b), the in-band signal The examples above are all tone signals, but by increasing the number of tone signals or modulating the tone to make it a subcarrier signal, it is possible to This makes it possible to transmit various types of signals at high speed.The above was a case of analog signals, but if a digital data signal is used as a control signal, the audio signal is also digitally encoded, and both can be transmitted at high speed. It is also possible to divide and multiplex the audio signal for transmission, and the circuit configuration in this case is shown in Figure 3D.
This is an example of a case in which the data signal is digitized at 1, multiplexed with a data signal by a multiplex conversion circuit 92, and applied to a modulation circuit included in the wireless transmission circuit 32. However, in digital data signals, there is usually no multiple load gain when multiplexing analog signals, which will be described later, so this point must be kept in mind when designing the system. Then, it is received by the opposing receiver, and the 3D signal is received by the demodulation circuit.
By performing the operation opposite to that shown in the figure, it is possible to extract the audio signal and the control signal separately. On the other hand, the signal sent from the mobile radio device 100 is received by the antenna section of the radio base station 30 and input to the radio reception circuit 35. FIG. 2A (b) schematically shows this upstream input signal. That is, time slot SU1. SU2. ...S
un is mobile radio 100-1.100-2°...,
100-n to the wireless base station 30 is shown. Also, each time slot SU1. SU2, ..., su
If the contents of n are shown in detail, it consists of a synchronization signal and a control signal or/and a telephone call signal, as shown in the lower left of FIG. 2A (b). However, if the distance between the radio base station 30 and the mobile radio FM 100 is small or depending on the signal speed, the synchronization signal may be omitted. Roughly speaking, the waveform of the radio carrier wave of the above-mentioned uplink radio signal within a time slot is schematically shown as shown in FIG. 2B (C). Now, among the input signals that have arrived at the wireless base station 30, the control signal is immediately sent to the control unit 40 from the wireless receiving circuit 35.
added to. However, depending on the magnitude of the speed conversion rate, it is also possible to apply the output of the signal speed restoration circuit group 38 to the control unit 40 after performing similar processing on the call signal. Further, the call signal is applied to the signal selection circuit group 39. A timing signal from a timing generation circuit 42 that generates a predetermined timing is applied to the signal selection circuit group 39 according to a control signal instruction from the control unit 40, and a synchronization signal and a control signal are generated for each time slot SU1 to Sun. The signal or speech signal is separated and output. Each of these signals is input to a signal speed restoration circuit group 38. This circuit is a speed conversion circuit 131 (first
It has a function to perform the inverse conversion of Figure B-1), which faithfully reproduces the original signal and replaces the barrier! It will be sent to fi20. The manner in which signals are transmitted in the signal space according to the present invention will be explained below using the required transmission band and the relationship between this and adjacent wireless channels. As shown in FIG. 1C-1, the control signal from the control unit 40 is sent to the radio transmitting circuit 3 in parallel with the output of the signal allocation circuit group 52.
Added to 2. However, depending on the size of the speed conversion rate, the signal allocation circuit group 5
It is also possible to add the output of 2 to the wireless transmission circuit 32. Next, in the mobile radio 1100: 1B-1
As shown in the figure, one of the functions of the wireless base station 30 is the communication path.
It has the circuit configuration required when it is used as a channel. Original signal, for example, an audio signal (0.3KHz to 3.0K
FIG. 3B shows the frequency distribution on the output side when I-1z> passes through the signal speed conversion circuit group 51 (FIG. 1C-1). In other words, if the audio signal is converted 15 times as described above, the frequency distribution of the signal will be 3B.
The figure has been expanded to < 4.5KHz to 115KH2. Although the frequency distribution of the signal has been expanded here, it should be noted that the waveform form has simply been stretched along the frequency axis, and the waveform itself has not changed. This is necessary when determining the value of multiple load gain. Now, FIG. 3B shows a case where the control signal is simultaneously transmitted using the lower frequency band of the audio signal. Among these signals, the control signal (0,2 to 4.0
KH2) and a speech signal CH1 (denoted as SDl in 4,5-45KH2) are accommodated in a time slot, for example SDI. The same applies to the audio signals accommodated in the other time throwers 1 to SD2 to S[)n. That is, time slot SDI (+=2°3,
. However, the signals in each time slot are arranged in chronological order, and the signals in multiple time slots are not applied to the wireless transmission circuit 32 at the same time. These call signals are sent to the wireless transmission circuit 32 along with control signals.
When added to the angle modulator included in the angular modulator, the required transmission band requires at least fo±45KH2. However, fo is a radio carrier frequency. If there are a plurality of wireless channels given to the system, the speed-up of the signal by the signal speed conversion circuit group 51 is limited to a certain value due to the limitations on these frequency intervals. The frequency interval of multiple wireless channels is f
rep and the maximum signal speed due to the above-mentioned speed increase of the audio signal is fH, then the following inequality must hold between the two. f > 2 f H ep On the other hand, in digital signals, audio is usually digitized at a speed of about 64 kb/s, so read by raising the scale on the horizontal axis in Figure 3B, which explains the case of analog signals, by about one digit. Although necessary, the relationship in the above equation also holds true in this case. Further, the control signal input from the mobile radio device 100 to the radio base station 30 is input to the radio reception circuit 35, but a part of the output is input to the control unit 40, and the other part is input to the signal selection circuit group 39. The signal is sent to the signal speed restoration circuit group 38 via the signal speed restoration circuit group 38. After the latter control signal undergoes a speed conversion (conversion to a low speed signal) that is completely opposite to that at the time of transmission, it becomes the same signal speed as that used in the general telephone network 10 and is transmitted via the signal processing section 31. Sent to barrier exchange 1120. FIG. 1B-2 shows the circuit configuration of mobile radio ta100B that communicates with radio base E30. The mobile radio R100B is suitable for simultaneously transmitting and receiving not only speech (telephone) signals but also image signals. The speech signal and image signal including the control signals of the two channels received by the antenna section are sent to the reception mixer 136-1 and the reception section 1, respectively.
37-1 and a radio receiving circuit 1 including a receiving mixer 136-2 and a receiving section 137-1.
35-2 and its output communication signal is input to two speed recovery circuits 138-1, 138-2 and a clock regenerator 141, respectively. The clock regenerator 141 regenerates a clock from the received signal and applies it to the speed recovery circuits 138-1 and 138-2, the control section 140, and the timing generator 142. The speed restoration circuits 138-1 and 138-2 restore the speed (pitch in the case of analog signals) of the compressed and segmented communication signals in the received signal, convert them into continuous signals, and send the control signal to the control unit 140. The ID signal is input to the ID information verification storage section 182, and the call signal and image signal are applied to the telephone section 101 and the image terminal section 102, respectively. Communication signals output from the telephone unit 101 and the image terminal unit 102 are transmitted through two speed conversion circuits 131-1 and 131-1, respectively.
131-2, the communication signal is divided into predetermined time intervals and its speed (pitch in the case of analog signals) is increased (compressed).
A wireless transmitting circuit 1 including a transmitting mixer 133-1 and a transmitting section 134-1 in order to transmit using two channels.
321 , a transmission mixer 133 - 2 , and a transmission section 1342 , the transmission signal is applied to a radio transmission circuit 132 - 2 including a transmission mixer 133 - 2 and a transmission section 1342 , and the transmission signal is sent out from an antenna section and received by the radio base station 30 . In the timing generator 142, the transmission/reception intermittent controller 123 . Speed conversion circuit 131-1, 13
1-2. It supplies necessary timing to the radio reception circuit 135-1, 135-2 and the speed restoration circuit 138-1 and 138-2. Further, the clock from the clock regenerator 141 is also applied to the speed conversion circuit 131-1.131-2. This mobile radio device 100B further includes a synthesizer 12.
1-1 to 121-4, changeover switches 122-1 to 122-4, and changeover switches 122-1 to 12
The synthesizer 121-1 to 121-4 and the transmitting/receiving intermittent controller 123 simultaneously transmit and receive two channels. It is controlled by the control unit 140 so that it can be executed. A reference frequency is supplied from a reference crystal oscillator 120 to each synthesizer 121-1 to 1214. 1
Not only can it transmit and receive using two time slots of one radio channel, it also allows frequency diversity by transmitting and receiving on two channels simultaneously. The ID information collation storage unit 182 uses the identification information (ID) transmitted from the wireless base station 30 to the speed restoration circuit 138-1.
, 138-2, and under the control of the control unit 140,
Check with the memory contents and memorize as necessary. Figures 1C-2, 1D-2 and 1D-3 include:
The overall configuration of the wireless base station 30B, which is suitable for transmitting and receiving communication signals including image signals, a specific example of the switch group 83 that is a component thereof, and the signal speed restoration circuit group 38. Signal selection circuit group 39° Signal speed conversion circuit group 51. A specific configuration example of the signal allocation circuit 152 is shown. N-channel communication signals 22-1 to 22- with the gateway exchange 20
n is connected to a signal processing section 31 forming an interface through a transmission path. Now, as shown in FIG. 1C-2, the communication signals 22-1 to 22-n sent from the barrier exchange 2O are input to the signal processing section 31 of the radio base station 30B. Signal processing section 31
In addition to being equipped with an amplifier to compensate for transmission loss, so-called 2-wire to 4-wire conversion is performed. That is, the input signal and the output signal are mixed and separated, and the barrier switch 20
The input signals from the switch 5RA-1-1, 5RA-1-2, . . . , 5R as shown in FIG.
A-1-n, 5RA-2-1, 5RA-2-2゜-, 5
RA-2-n, -=-, ., 5RA-n-1. 5RA-rl-2, 5RA-n-n, and 5RB-1
-1,5RB-1-2,-,5RB-1-n. 5RB-2-1, 5RB-2-2,..., 5RB2-
n,-,-,5RB-n-1,5RB-n-2,-,5
RB-n-n, and 5TA-1-1゜5TA-1-2
, , 5TA-1-n, 5TA-2-1, 3TA-2-
2,·, to ST-2-n. +++, +++, Scho A-, n-1, 3TA-rl-2
,-. 5TA-n-n, and 8TB-1-1, 8TB-1-2,
-, 8TB-1-n, 5TB-2=1. STB-2-2
,-,3TB-2-n, ・, ・,3TB-n-1,
5TB-n-2, -, 5TB-nn, the signal speed conversion circuit groups 51A and 51B (first
(see Figure D-3). Further, the output signals from the signal speed restoration circuit groups 38A and 38B, the details of which are shown in FIG. 1D-3, are sent to the signal processing unit via the switch group 83 as shown in FIG. 1D-2 and FIG. 1G-2. At step 31, the signal is transmitted to the barrier exchange 20 using the same transmission path as the input signal. Here, switch 8\83 is the transmission switch 5TAI-
1~S block A-n-n, 3 block B-1-1~S block [3-n-
n and reception switch 5RA-1-1 to SR△-n-
n, 5RB-11 to SRB-ml-n,
All of them are controlled by the communication path control unit 81 to open and close the switch group 83 to achieve the desired purpose, and operate to enable transmitting and receiving diversity. The ID identification storage section 82 is used to identify and store the ID of the mobile radio device 100. In addition, the component 81 with a communication path opens and closes the switch group 83 according to commands from the control unit 40 to control the communication path, but the communication path control unit 81 also provides information and requests for control to the control unit 40. It has the function to do. Among the above, the input signal from the barrier exchange [20] passes through the switch group 83, and then passes through many signal speed conversion circuits 51-1.
~51-n coarse signal speed conversion circuit group 51
A and 51B, and are divided into predetermined time intervals and subjected to speed (pitch) conversion. Also, the signal transmitted from the wireless base station 30B to the gateway exchange 20 is transmitted to the wireless receiving circuit 35A.
, 35B are sent to signal speed restoration circuit groups 38A and 3 through signal selection circuits u39A and 39B, respectively.
After being inputted to 8B and subjected to speed (pitch) conversion, it is inputted to the signal processing section 31 through a switch group 83. Now, the control of the radio receiving circuits 35A and 35B or the output of the speech signal and the image signal are input to signal selection circuit groups 39A and 39B, respectively, each including a set of signal selection circuits 391 to 39-n that select signals for each time slot. Here, speech signals and image signals are separated corresponding to each communication channel CH1 to CHn. This output is provided by the signal speed restoration circuits 38-1 to 38-3 set for each channel.
After the signal speed (pitch) has been restored by the signal speed restoration circuit groups 38A and 38B each including the groups 8-n,
It is input to the signal processing unit 31 via the switch group 83, and the 4
After undergoing line-to-wire conversion, this output is sent to the barrier switch 20 as communication signals 22-1 to 22-n. Next, the call originating/receiving operation of the system according to the present invention will be explained by taking the case of a voice signal as an example. (1) Call origination from mobile radio device 100 This will be explained using the flowcharts shown in FIGS. 4A and 4B. When the power of the mobile radio device 100 (the same applies to the mobile radio device 100B) is turned on, the radio receiving circuit 135 in FIG.
) Start capturing the control signal included in the wireless channel (channel CH1). If the system is provided with more than one radio channel, the radio channel exhibiting the highest received input electric field; i) the radio channel indicated by the control signal contained in the radio channel; i) the time interval within the radio channel; A radio channel (hereinafter referred to as channel CHI), such as a channel with an empty time slot among the slots, enters the receiving state according to the procedure determined by the respective system. This is possible by capturing the synchronization signal within the time slot SDi shown in FIG. 2A(a). In the control unit 140, the synthesizer 121-
A control signal is sent to the synthesizer 122-1 to generate a local frequency that enables reception of the radio channel CH1, and the switch 122-1 is also fixed in the position of the synthesizer 121-1. Therefore, when the receiver of the telephone unit 101 goes off-hook (starts making a call) (S201, FIG. 4A), the synthesizer 121-2 of FIG. A control signal is received from the control section 140 to generate the following. Further, the switch 122'-2 is also turned toward the synthesizer 121-2 and becomes fixed. Next, the calling control signal output from the telephone unit 101 is sent out using the wireless channel CH1. This control signal is controlled by the frequency band shown in FIG. 3A(b).
For example, it is transmitted using a time thrower +-sun. The transmission of this control signal is limited to time throwers 1 to SUn, and is sent in bursts, and no signal is transmitted during other time periods, so that it does not adversely affect other communications. However, if the speed of the control unit g is relatively slow or the amount of information of the 13 bow is large, one time thrower 1
If it cannot be accommodated within ~, it is transmitted one frame later or further using the same time slot in the next frame. Specifically, the following method is used to capture the time slot Sun. As shown in FIG. 2A (a), the control signal transmitted from the radio base station 30 includes a synchronization signal and a subsequent control signal, and the mobile radio device 100 receives this signal to perform frame synchronization. becomes possible. Furthermore, this control signal includes control information such as currently used time slots and unused time slots (empty time slot display). Depending on the system, the time slot SDI (1=1.2°...
n) is used by remote communications, it may contain only synchronization and speech signals, but even in these cases unused time slots usually contain synchronization and control signals. By receiving this control signal, mobile radio 11100 can know which time slot to use to send out the calling signal. Note that if all time slots are in use,
It is not possible to make a call on this radio channel, and it is necessary to sweep and search for another radio channel. In another system, there may be no empty slot indication in any of the time slots, and in this case, the following audio multiplex signals SD1, SD2, . ..., S
It is necessary to search for the presence of Dn one after another and check for empty time slots. Now, returning to the main topic, the mobile radio device 100, which has received the control information sent from the radio base station 30 by one of the above methods, determines in which time slot the mobile radio device 100 should transmit the call control signal. It is possible to make a judgment including the transmission timing. Therefore, assuming that the time slot Sun for uplink signals is an empty slot, it is decided to use the empty time slots 1~, and the control signal for calling is sent out, and the necessary timing is determined from the response signal from the wireless base station 30. It is possible to extract a burst control signal and send out a burst control signal. If there is a call from another mobile radio at the same time, the call signal will not be properly transmitted to the radio base station 30 due to call collision, and the operation will have to be restarted from the beginning, but this probability is When viewed as a system, this value is kept to a sufficiently small value. If call collisions are to be significantly reduced, the following method may be used. If the mobile radio 1100 finds an empty time slot in which it can make a call, it does not use the entire time slot, but uses only the first half for some mobile radios and the latter half for others. There is a way to do it. In other words, the time signal is used as a calling signal.
Divide the used portion of the slot into several types, use this to classify a large number of mobile radio devices into groups, and place one of the slots in each group.
This method gives the time period within one time slot. Another method is to create many different frequencies for the control signal,
This is a method of dividing a large number of mobile wireless devices into groups and applying this to each group. According to this method, even if control signals of different frequencies are transmitted simultaneously using the same time slot, no interference will occur at the radio base station 30. The above two methods may be used separately, or when used together, the effects will increase synergistically. Now, suppose that the call control signal from the mobile radio device 100 is successfully received by the radio base station 30 and the ID (identification number) of the mobile radio device 100 is detected (S202). Search for time slots. The time slot given to the mobile radio 11100 may be Sun, but a search is performed just to be sure. This is done in order to respond to simultaneous calls from other mobile radios in addition to the mobile radio 100, and to provide a time thrower 1 suitable for the service type or service classification. As a result, if the time slot SD1 is found to be empty, for example, the time slot SD1 is assigned to the mobile radio device 100 using the time thrower l-SD1 of the radio channel Up (mobile radio 1
00→Radio base station 30) SUl. and instructs to use the corresponding downlink (radio base station 30→mobile radio device 100) SDl (S203
>. In response, the mobile radio device 100 transitions to a state in which it can receive data in the instructed time slots 1 to SD1, and also changes the time slot 5U1 (5U1), which is the uplink radio channel time slot corresponding to the downlink time slot SD1. 2A
(see figure (b)). At this time, the mobile radio device 10
In the control unit 140 of 0, the transmission/reception intermittent controller 123
, and switches 122-1 and 122-2 start operating (3204). At the same time, it sends a slot switching completion report to the radio base station 30 using uplink time slot SU1 (3205>, and waits for dial tone (3206>. Time slot SU of the radio carrier of this uplink radio signal).
The state of No. 1 is schematically shown in FIG. 2B (C). In addition to the time slot SU1, the radio base station 30 receives an uplink signal SU3 from another mobile radio 1100.
Ya Sun is included in one frame and sent. The radio base station 30 that has received the slot switching completion report (
S207>, sends a calling signal to the barrier switch 20 3208>, and receives this

[The puta barrier exchange 20 detects the ID of the mobile radio 100 and turns on the necessary switches among the switch groups included in the barrier exchange FM 20 (3
209>, sends a dial tone (3210, Figure 4B). This dial tone is transferred by the radio base station 30 (S211>, and the mobile radio 1100 confirms that the communication path has been set (S212>.
Since a dial tone is heard from the 01 handset, the dial signal begins to be sent. This Daicel signal is speed-converted by a speed conversion circuit 131 and sent to a transmitter 134. P3 and the wireless transmission circuit 132 including the transmission mixer 133 using the uplink time slot SU1 (S213>
. Thus, the transmitted dial signal is sent to the wireless base station 30.
It is received by the radio receiving circuit 35 of. This wireless base station 30
Now, in response to the calling signal from the mobile radio device 100, the time slot to be used is given, and the signal selection circuit group 39 and signal allocation circuit group 52 of the radio base station 30 are operated to determine the uplink time. - It has received the slot SU1 and is now in a state of transmitting the signal of the downlink time slot SD1. Therefore, the dial signal transmitted from the mobile radio device 100 passes through the signal selection circuit 39-1 of the signal selection circuit group 39, and then is input to the signal speed restoration circuit group 38, where the original transmission signal is restored. The communication signal 22-1 is sent to the gateway exchange 2 via the signal processing unit 31.
0 (S214>, and a call path to the telephone network 10 is set (3215>). On the other hand, input signals from the barrier switch 20 (initial control signal,
When a call is started, the call signal (call signal) undergoes speed conversion in the signal speed conversion circuit group 51 at the wireless base station 30.
The time slot SDI is provided by the signal allocation circuit 52-1 of the signal allocation circuit 152. Then, the time slot S of the wireless channel downstream from the wireless transmission circuit 32
It is transmitted to the mobile radio device 100 using D1. The mobile radio R100 is waiting for reception in the time slot SD1 of the radio channel CH1, and is received by the radio receiving circuit 135, the output of which is input to the speed restoration circuit 138. In this circuit, the original signal of the transmission is restored and input to the handset of the telephone section 101. In this way, a call is started between the mobile radio 11100 and a regular telephone in the general telephone network 10 (3216>. To end the call, put the handset of the telephone unit 101 of the mobile radio 1100 on-hook. (3217), the call end signal, the on/knock signal from the control unit 140, and the speed conversion circuit 1
31 from the wireless transmitting circuit 132 to the wireless base 830 (3218>), and the control unit 140 stops the operation of the transmitting/receiving intermittent controller 123 and switches the switches 1221 and 122-2 to the synthesizer 1.
It is fixed to the output ends of 21-1 and 121-2. On the other hand, in the control unit 40 of the radio base station 30, the mobile radio device 1
When the call termination signal g from 00 is received, the call termination signal is transferred to the gateway exchange 20 (5219), and the switch 8 (not shown) is turned off to terminate the call (S220). At the same time, the signal selection circuit group 39 and the signal selection circuit group 52 in the radio base station 30 are opened. In the above explanation, control signals are exchanged between the radio base station 30 and the mobile radio R100 by the signal rate conversion circuit δY51.
Is it explained that there is no signal speed signal speed restoration answer group 38? This is for convenience of explanation, and like the audio signal, the signal speed conversion circuit 8 ¥51, the signal speed restoration circuit group 38, the control signal speed conversion circuit 48, etc. Communication can be performed through the signal processing unit 31 without any problem. (2) Incoming call to mobile radio device 100 The mobile radio device 100 is on standby with the power turned on. In this case, as explained in the section regarding the call origination from the mobile radio device 100, the mobile radio device 100 is in a waiting state to receive a downlink control signal of the radio channel CH1 in accordance with the procedure defined by the system. Assume that an incoming call signal from a general telephone m10 to the mobile radio device 100 arrives at the radio base station 30 via the barrier switch 20. These control signals are transmitted as communication signals 22 to the control unit 40 (FIG. 1C-1) via the signal speed conversion circuit group 51 and the signal selection circuit group 52, similarly to the voice signals. When the mobile radio device 100 is pressed, the control unit 40 uses an empty slot in the downlink time slot 1 of the radio channel CH1 addressed to the mobile radio device 100, for example, SDl, to display the ID signal of the mobile radio device 100, the incoming call signal display signalman time. - Slot 1 - Use signal (for example, SD
(using SUl corresponding to l). In the mobile radio device 100 that receives this signal, it is transmitted from the receiving section 137 of the radio receiving circuit 135 to the control section 140. Control unit 14
0, it is confirmed that this signal is an incoming call signal to the own mobile radio 100, so the telephone unit 101 makes a ring tone and at the same time calls the designated time slot S.
D1. Transmission/reception intermittent controller 123 to standby with SU1
In addition to operating the switches 122-1 and 122-
2 starts turning on and off. In this way, the state has shifted to a state in which calls can be made. It should be noted that it is theoretically explained in Reference 2 that signal transmission is carried out in good condition using this system and that it does not adversely affect other wireless channels in the system, so it will be omitted here. The fact that the TCM signal applied to the present invention has multiple load gain will be explained theoretically, and then its application will be described. (3) Regarding the multiple load gain of the TCM signal transmitted from the wireless base 830, in order to correlate the multiple load gain of the TCM (time minute υ1 time compression type) signal with the multiple load gain of the FDM (frequency division multiplex) signal, First, each channel CH of FDM
1, C1-12, and -CHn are expressed in the form of a function. As is well known, FDM signals are frequency-converted audio signals and arranged in a line on the frequency axis as shown in Figure 5. This multiplexed signal is often used in coaxial transmission systems and microwave analog communication systems. , and multiple load interest (
q has also been incorporated into practical systems and is showing great effects. Hey, the spectrum in Figure 5 has 1 channel.
The case of 2 pieces (CH1 to 12> is shown, but in general, 1 piece is used.
2 others 60,120,480゜960.1200.2
A wide variety of 700 pieces are used. Now, the channels CH1, CH2, CH3, ---, C in FIG.
Audio signal flowing to l-1n (In the case of wired, the frequency band to be transmitted is 0.3 to 3.4KH7, but for mobile radio signals, it is 0.3 to 3.OK Hz, so it is limited to this value. ) as fl(j), f2 (t), -・-・,
Let f, (t). The right frequency components of these signals are 0.3 to 3.0 KHz for channel CH1,
Cl-12 is 4.3-7.0KHz. --・-・-, CHn or 4x (n-1) + 0.3
~4x (n-1)Kl-12, and there is no overlap with each other. However, when viewed from the signal waveform, the amplitude valve m of these ng audio signals is simply shifted to a higher frequency on the frequency axis, and the signal g waveform itself does not change at all. This is important in determining the multiple load gain, and can be expressed as follows. The known multiple load gain jq of the FDM signal is exactly the same as the signal when n voices are arranged on the frequency axis as shown in Figure 5, and the signal when n voices are simply mixed without frequency conversion. . Prove this using a formula. The mixed signal of channels CH1, Cl-12°, . . . , CHn is expressed by the following equation. F(t) −f (t) 10f2 (t) 10...10f. (1) (1) Specifically, f and (t) are expressed as follows. kHz (2) (4i-1)kHz (3) However, ≧2 In addition, the mixed signal when frequency conversion is not performed is given by the following equation. G(t) g1 (1) + Q 2 (1) +-+ g n (1) (4) Here, gl (j) = f1 (t) kllZ (5) However, 1≧2 Then, ( 1) Find the power of the signal in equations (4). First, ``The power of (t) is 2 +f2(t) +-+fn(t) (7) On the other hand, the power of G(1) is d[-Ω1 (1) 10g2 (1) However, different Between the signals, we used the following. That is, +...+Q, (t) (8) No power is formed 1g gjdt=Q However, 1≠" Now, equations (2) and (3) From (9) (10) (11> However, 1≧2 (10), from equations (11), F(t) ” 30(t) 2(12) In other words, the power of the signal is It became clear that it is not related to frequency conversion.Let us consider applying the sampling theorem to q・(1).Here, since the highest frequency fh of J(t) is 3KH2, the time interval 1/ (2fh), that is, every 000 seconds of one melody. , f・(1) as shown in FIG. 6A (a), each time interval E 1/6000+(1/6000)(i -1)/6
000] seconds (13). In the same figure, t 8 = 1
/6000 seconds. ↑b-(1/6000) x (1/6000) seconds. to= (1/6000) x (5999/6000
) seconds. td=1/6000+(1/6000) x (300
0/6000) seconds. t o = 1/6000 seconds. It is. Hereinafter, for specific explanation, the multiplex number n is set to 8.
000. The length of one frame is 1/6000 seconds. Now, on the horizontal axis of Figure 6A (a), put 600 small bags (diameter 1/6000 x 1/6000 seconds) as shown in Figure 6B (c).
0 pieces, 1 g of large bags with a diameter of 1/6000 seconds, are arranged as shown in the figure. Then, compare the above sampled values to matchsticks and consider how these matchsticks fit into the bag. Since the function q・(t) (:=1.2..., n) has 6000 matsushiko each in one second, and they are equally spaced in time, the capacity is (1). , [2),...,(
One frame is sent to each frame (N), and this operation is repeated for each frame. That is, Bag (1) has Ql (1/6000) Bag (2) has g2 (1/6000 + 1/6000x
1/6000) bag (3) contains g3 (1/6000+
g (, 1/6000+ 1/6000000 x 5999 /6000) will be placed in each of the 1/6000x 2/6000) bags (6000). Also, in the large bag (Σ6000), the mixed signal G(t)
I decided to include a Matsuchi stick that was sampled. In this case, the sampling time is 1/6000 + 1/
6000x3000/6000, that is, the midpoint of one frame. Then, the value of the next match salary will be stored in the large bag (Σ6000). For large bag (Σ6000>, G (1/6000+1/6000x 3000 /6
000) Now, below are the values of the matchsticks for bags (1) to (6000), Q 1 (1/6000) '; J 2 (1/6000 + 1/6000x 1/
6000). Q 6000 (1/6000+1/6000x 59
99/6000) and the large bag (Σ6000),
The value of the Matsushi offering G (1/6000 + 1/6000x 3000 /
6000) are compared as shown in equations (14) and (15) below. 000 10g2 (1/6000+ 1/6000 mowing/600
0) 10g3 (1/6000+ 1/6000X
2/6000) 1/6000+1/6000
x (1-1)/6000 )+q6000 (1/
6000+1/6000x 5999 /6000)(
14> G (1/6000+1/6000X3000/600
0 ) =Q1 (1/6000+1/6000x3
000/6000 )+g2 (1/6000+1/
6000x3000/6000 )+・・・・・・ +gn(1/6000+1/6000x3000/60
00 ) (15) As a result of comparing equations (14) and (15), if +Δ (16) and Δ converges to its average value or O every time it is sampled at 1/6000 second intervals, then , it has been proven that the multiload gain in the FDM signal is similar and has the same value in the TCM signal. This is because 6000 bags placed on the horizontal axis have a capacity of 1 bag.
It represents the time thrower 1~, and the sum of the bags is hJ or 1 frame, and the matsushiko in the bag is each signal Q, g,
, g or hour/minute 12°""'n It can be considered as a time compressed multiplexed signal (DCM>), and the DCM signal is well mixed as shown in the large bag (Σ6000) in Figure 6B (C). This is because, in this example, even though it is called a CM signal, there is no particular need for time compression (,11,
The compression degree must be 1. By the way, in Figure 6B (C), the position of the bag on the horizontal axis (assistance axis) is as follows: Bag (1) is 1/6000≦↑< 1/6000+1/6000x
1/6000 bag (2) is 1/6000+1/6000x 1/6000≦t<
1/6000+ 1/6000x 2/6000 Bag (i) is 1/6000+ 1/6000X (i-1)/600
0≦t<1/600011/6000x i/60
00 bags (6000) are 1/6000+ 1/6000x 5999/6000
≦t < 1/6000 1/6000x 5999/
6000 is installed. On the other hand, the audio signal Ω・(j) , Q2 (j) ,..., g
The sampling time of i(t),...g6000(t) is 1/600, respectively.
0.1/6000+ 1/6000x 1/6000.
......, 1/6000+ 1/6000X (
i-1)/6000. ......, 1/60
Since it is set to 00+1/6000x5999/6000, each matsushi stick will enter the one-piece bag while touching the side of the bag. This is done for convenience; if you want to put a matsushigi in the center of the bag, for example, change the time t of q・(1) to t=1/6000+1/6000x(i+0.5)/
6000 may be selected. However, even with this selection, the conclusion of this proof does not change. Now, compare the corresponding terms on the right sides of equations (14) and (15). △・ -q・ (1/6000+1/6000X (1
-1)/6000)g(1/6000+1/6000
x3000/6000) (17) What the above formula means is that when sampling the audio of the first channel (CHl), the time (1/6000 and a half [1-1)/60002) seconds,
(1/'6000+3000/60002> seconds) This difference is like the error value of random noise, and the channel (CHi) i = 1, 2.-, n In,
It may take a positive value, a negative value, or occasionally O, but generally it will vary from side to side with O as the center, and the variation will be normally distributed. The above was about a certain time ↑=1o. Each sampling takes place after 1/6000 seconds. The next one is delayed by another 1/6000 second. For example, this sampling is performed 6000 times per second, so the average value of equation (17) for one second is 000 Mean (Σ Δ, ) = 1/6000 ΣΔ, →
O(18), that is, it will approach O. This will become clearer if more time is spent, and it can be considered that equation (18) becomes O if the average is taken over 10 seconds or 30 seconds. As described above, FDM and T with the same multiplicity (6000)
It has been clarified that the average power level of each CM signal is the same, and next we will prove that the amplitude distributions of the two types of multiplexed signals are also exactly the same. When the number of FDM signals reaches 60 or more, the waveform of the signal becomes almost the same as random 'a@ (also called white noise because the amplitude characteristics are flat in the target frequency band). is well known. Now, considering the above-mentioned equation (4), this means that as shown in equation (1), if FDM (not No. 8, but 60 call signals or more) is multiplexed, each signal is a random signal. It can be considered as a signal that is a mixture of 60 signals. Therefore, the signal of formula (4) is
). (1/6000+1/6000x 9000/6000
). (1/6000+1/6000x(3000xk)/6
000) (k=1.2, 3......) and the signal group obtained by sampling at
Considering the amplitude distribution of (referred to as a group), 1υ
The value of the signal is obtained by sampling a random signal with a constant width at regular intervals (k = 1.2, 3, etc.), so it is clear that It can be seen that this amplitude distribution remains flat regardless of time. On the other hand, the signal g1(t) shown on the right side of equation (4)
, g2 (t) ,..., CI,, (t
), the amplitude distribution of the signals sampled at the times shown by equation (13) is still random, and it will be proven below that the magnitude of the amplitude is the same as that of equation (15). Random signals g1 [t), g2 (t),...
...,q. The temporal distribution of equation (14), which is a mixed signal obtained by simply changing the sampling time of each of (1), is taken. In other words, create a group (hereinafter referred to as group B) of signal values sampled at each time given by the formula (14) below. 000 10 2 (k/6000 + 1/6000x 1/6
000)+・・・・・・ 10g 6ooo(k/6000 +1/6000x 5
999/6000) (14-a> (k = 2.3, 4...) Amplitude distribution of the value 000 of a large number of signals that is (q) in equations (14) and (4-a) Considering that, even if the sampling time of the random signal is changed, the shape of the distribution does not change, and therefore the amplitude distribution of the signal group in equation (4) is exactly the same.From the above, the FDM signal (however, in this case It has become clear that a multiplexed signal (such as equation (4) that is not expanded on the frequency axis) and a CM signal with the same multiplicity have the same average power and the same width. The above means that the iq for each multiple load obtained by FDM is T
It just goes to show that you can get what you want from commercials as well. However, the value of the multiple load gain quoted from Document 3 is the audio signal frequency band, or 0.3 to 3.4KH2, whereas It was assumed that the same value could be obtained in the transmission band of 0, 3 to 3.0 KH2. This assumption can be accepted with virtually no error. In addition, in the case of a CM signal, since the signal is time compressed,
The frequency components it has become higher by the degree of compression, but this is because only the frequency components have been changed as mentioned above, and the signal waveform itself has simply been extended on the frequency axis, so the multiple load gain ( Although this does not change, we will prove it strictly using mathematical formulas below. The CM signal can be expressed as follows using formulas (5) and (6): 3.0KHz (19) However, T<t<T/n+eT h・(t)=0 However, 1/n<t!T<t<d+2t/=1.2,
3.・・・・・・ The time length of one frame is shown below. Therefore, the time-compressed CM signal is 1β ・
))Xn1/2 p (20) The reason for multiplying by n1/2 on the right side of equation (20) is that the TCM signal is transmitted only for 1/n of the time within one frame. This is expressed in terms of voltage (1 times the power). Now, if we calculate the power of equation (2o) for the time T of one frame, we get d1f l-1(t) 2dt- -n/T [T/rlΣb ], similar to equations (7) and (8). (21) Therefore, if the time is an integral multiple of the frame, the signal H
(t) and G (t) are found to be 1-1(t)230ft')2 (22>).Next, the frame length of the audio n-channel multiplexed TCM signal is 1
/(2fh). In this case, as above, the FDM of voice n-channel multiplexing is
It can be easily proven that it has a value equivalent to the multiload gain in the signal. For example, assume that the frame length is 1/8000 seconds. Then, the sampling frequency was changed from 6000112 to 8000Hz, and each audio signal gg1 (t),
Cl3(t),...2g, (1) is sampled, and the mixed audio signal G(t) is also sampled as 80001-(Z
All you have to do is sample them and compare them. That is, Fig. 6A (a>, (b), Fig. 6B (
All of the above explanations can be applied by simply changing the horizontal axis of C) from 1/6000 to 1/8000. Briefly, we will explain what happens to the multiple load gain when the frame length becomes longer than 1/(2fh>.The conclusion is that the multiple load gain generally decreases, but the specific value is The calculation is as follows.As a concrete value, the number of multiplexes is 6000, which is the same as the previous example, and the frame length is 1/3000.
This will be explained by taking as an example a case in which the number of seconds is reached. The size of the bags arranged on the time axis increases by the increase in one frame length, as shown in FIG. 6B (d). To be exact, the diameter of the container is 1/6000 x 1/3000 seconds. Then, if we add up the diameters of all n bags, we get 1/3000
seconds. Also, the large bag (Σ6000) is 1/30 in diameter.
00 seconds. Now, as above, the sampling frequency is 17600.
Consider putting the Matsutchi stick sampled at 0 seconds into a container. In this case, if the frame length is set to 1/6000 seconds as described above, the following inconvenience will occur. That is, FIG. 6B (d
) As shown in 1/6000 seconds, the bag is (1) ~
(3000), while Matsushibo has 6000.
Since I have books, I can put two books in each bag. That is, as shown in FIG. 6B (d), bag (1) has audio signals g1(t) and g2(t), and bag (2) has audio signals g2(t) and g2(t).
t) and q3 (t), and bag (3) has g(t) and g4(
1),..., below, the bag (3000) is Ω3000 (
Matchsticks (signal values) indicating 03001 (t) and 03001 (t) are entered. Note that gl(t) is placed in a bag (6000) which is the previous sampling time. On the other hand, bags (3001) to (6000) do not contain any Matsushi sticks sampled within this time, and only 2 Matsushi sticks sampled within the next sampling time of 176,000 seconds are stored in the bags (3001) to (6000). Kien will be admitted. On the other hand, the large bag (Σ6000) has two matchsticks in one frame, so two matchsticks, that is, G (1/6000 + 1/6000 x 3000/600
0) and G (1/6000+1/6000x9000
/6000) will be inserted. However, G (1/6000+ 1/6000X9000/6
000 ) Lt is a matchstick (signal) sampled within 1/6000 seconds as described above. What kind of goods does the above refer to? That is, the fact that two matchsticks are placed in bags (1) to (3000) means that two channels of audio Q・(t) and ``i+i(t) are mixed in each time slot of the TCM signal. This means that the FD of 2 chitonels (CI-1>
M, total of 3000 slots 1 to 2 (C11)
3000 = 6000 (CH) Wang CM (means that an old name should be created. And these and Obukuro (Σ6
The size will be compared with the one Matsushibo in the front of the 000) in the same way as the previous) small. Although this is also a method, it is not a commercial signal in the original sense. Therefore, in order to include only one audio signal in each time slot, the following must be done. The 6000 audio signals are divided into 2u, so that one group can be placed in bags (1) to (3000), and the other group can be placed in bags (3001) to (6000). This operation will be explained using FIG. 6D (f). There are bags (1) to (6000) and large bags (Σ6000).
Two sets of each are available. Now, in the bags (1) to (6000) written at the top, there are matsushi sticks Q, Q, Q5. ..., q5999 is 1 each
3 It shows a man being put in a tree. And a large bag (Σ
6000) is G 1 (1/6000+1/600
There is one match stick showing 0x 3000/6000). On the other hand, the bags (1) to (6000) written at the bottom contain Matsushi-Ken Q2. g4. C]6. ..., ``6000 is put in one tree brother each, and G2 (1
/6000+ 1/6000X 9000/6000
) is included in the book. If you do this as shown in this diagram, you will not have two match sticks mixed together in the same bag, and the length of one frame, which I have already explained, will be reduced to 1/6000.
By the same proof as in the second case, we can see that the multiple load gain of the FDM signal (in this case, 3000 multiplexes) is TC
It can be seen that it is the same as that of the M signal. The above explanation can be expressed as follows. (4) n voices (in this case n-600
0) is divided into two and expressed as follows. al(t)=C]1ft)+Q3(t)10...10"5
999"(4') G2 (j) = g2 (1) + Q 4 (t) + ・=
+C76ooo(t)(4") Then, the above-mentioned proof can be performed for each of the above equations. However, the sampling timing is assumed to be performed under exactly the same conditions as in the previous case. In the above explanation, , it has become clear that there is a need for grouping in order to obtain the multiload gain, but it will be explained below that the TCM signal requires signal compression. In Figure 6D (f) It is true that the upper bag group (1) to (6000) and the lower bag group (1) to (6000) contain only 1 Mizuyuan of Matsushiko, but when viewed as a TCM signal, it is still unsatisfactory. The upper bag group and the lower bag group are in transit at the same time, so within one time slot of the DCM signal,
Still gl and g2゜g3 and C4・°゛°・” 59
99 and 06000 are supposed to coexist and become 0. Signal compression removes this, and the following method may be used. That is, gl, Cl3. ...1g59
Regarding 99, there are two bags (1) and (3001), (
2) and (3002), ..., (3000) and (60
00), respectively, to the previous bags (1), (2),...
., (3000), Cl3. g4. ...
Regarding 1g6000, do the same with the next bag (3001
), (3002), ..., (6000). To do this, for example, the following composite signal may be created for gl. That is, it is sufficient to create the sum of q signals at two adjacent sampling times. Technically, the audio signal is stored in a memory circuit and read out in a time-division manner at twice the speed (duty ratio 50%), and the signal obtained by sampling this read signal at a sampling rate of 1/3000 seconds is It is desired. However, in this case, faithful reproduction of the original signal is not possible with the instantaneous value of the CM signal (voltage value) of the sampling time, and it is necessary to transmit a signal with a fixed time width (time slot length). . Viewing the above operation from the viewpoint of multiple load interest) q, it is as follows. If the frame length is longer than the sampling time interval of 1/6000 seconds, even if the multiplex number n is 6000,
0C11 multiplex FDM a; multiple load publication) q is 1q
Therefore, if the frame length is 1/3000 seconds, the multiple load gain of the FDM signal is isometrically 3000 CL1. If the frame length is further increased to 1 second and the number of multiplexes is n = 6000, calculate the multi-small load benefit of 1 lb. An explanatory diagram in this case is shown in FIG. 6C (e). To explain the figure, one frame is one second, and n = 6
000, the diameter is 1/600 on the horizontal time axis.
There are 6,000 0-second bags, which form one frame. In this case C11(t),...
, q6000(t), respectively, and consider where they should be put. If the sampling timing is the same as in the previous case where the frame length of E is 176,000 seconds, then bags (1), (2), (3)
. . . . (6000), each of which represents each audio signal will contain one tree. On the other hand, in the large bag (Σ6000), there are 6000 matchsticks (having values sampled at each sampling period) indicating G(1). This means that T
This means that 6000 audio channels are mixed and input into each time slot of a CM signal. The technology that makes this possible is FDM, but this does not allow for TCM, which is the purpose of this application.
Cannot be applied to signals. Therefore, in order to have only one audio signal in each time slot, 6
000 audio signals are divided into 6000 groups, and each group is divided into (
In this case, it is necessary to input one channel at a time. This means that the multiple load gain in this case is O, which is completely 1.
It shows that you can't be ignored. Also, in this case, Ding CM
The signal compression degree of the signal is 6000. As is clear from the above explanation, we have shown that the multiple load gain decreases when the time length of one frame becomes longer than the sampling period required to faithfully reproduce the audio signal. If we use the standard, the frame length Tt becomes T
If t > 1/(2fh) and the number of multiplexes is n, the multiple load gain is n' = nx1/(2fhTo
> (23) This value is equal to the multiple load gain of a frequency division multiplexed signal having a multiplex number determined by the value. Frame length T1. According to the above-mentioned documents 1 and 2, the practical range of the multiplexing number n is as follows: Frame length T: 0.1 seconds≧Tt≧0.001
It is clear from the above example that as long as the second is the multiplex number n: 3000≧n≧2, and within the above range, equation (23) always holds true. In addition, in order to address one matchstick in the bag as a CM signal, in other words, to input 1 g of the original audio signal, the time length Tt of one frame must be 1,'' 2 f.
It has also become clear that if the signal is longer than h, the signal must be compressed, and the compression ratio ψ is given by ψ=Tt/(1/2h>).The above explanation applies only when the multiplicity is 60 or more. We have assumed this, but we will explain the case below.In this case, the number of bags in one frame in a CM signal is 60 or less.On the other hand, in an FDM signal, the multiplicity is 60 or less, so the signal can be considered as random noise. In other words, although the FDM signal or the signal shown on the right side of equation (4) may be said to be a random signal approximately, an error will occur in the result obtained as a random signal.However, In fact, whether or not it is a random signal has nothing to do with the multiple load gain, and the 2
The method that describes the characteristics of two signal value groups A and B is
Here, it is sufficient to apply the method again to the case where the multiplicity is 60 or less and prove that Au and the 8 groups have the same signal magnitude and distribution. First, the fact that the average amplitude (power) of group A is equal to that of group 8 can be determined by the same derivation from the sampling theorem already explained.Next, regarding the amplitude distribution, Again, since the audio signal does not have a certain periodicity, there is no reason why there would be a difference between the amplitude distribution of drunkenness and that of Bu. From the above, when the multiplicity is 60 or less, The multiple load gain of 41 for the CM signal is equal to the multiple load gain of 17 for the FDM signal with fffi] - multiplicity.However, when calculating the multiple load number 1q, the TCM signal multiplicity of 0 is given by equation (23). (4) Regarding the multiple load utilization 11 of TCM signals received at the radio base station 30, the radio base station 30 receives the TCM signals transmitted from a large number of mobile radios 100. However, let us consider the multiple load gain of this received wave.The conclusion is that, as will be described later, the mobile radio 100 has exactly the same multiple load gain as when transmitting from the radio base station 30. As a result, it can be seen that it is possible to transmit with a deeper modulation degree.As a concrete example, one frame length is divided into a sampling time interval of 1
/6000 seconds and the number of multiplexes is 6000. Wireless base station 3
0 is communicating simultaneously with 6,000 mobile radio devices 100 using the same carrier wave and using all of the time throwers 1 to 6,000 of one frame. The positions of the mobile radio devices 100 are equally spaced on the same circumference from the radio base station 30, and the receiving antenna of the radio base station 30 is omnidirectional, and the transmitting antenna of the mobile radio device 100 is also non-directional. The directionality and the magnitude of the transmission power from each mobile radio device 100 are all the same, and each mobile radio device 10
It is assumed that the carrier waves used for transmitting 0 are phase-synchronized with each other. Furthermore, the radio wave propagation characteristics between the mobile radio device 100 and the radio base station 30 are
0 and the radio base station 30 are the same. Under the above assumptions, the transmitted signals from each mobile radio device 100 that enter the radio base station 30 will be received exactly the same. Therefore, the appearance of the received signal within one frame in this case can be considered to be exactly the same as when it is transmitted from the wireless base station 30. Conversely, even though each mobile radio device 100 is transmitting only a single audio channel in the time slot given to it, the number of multiplexes is 6000 assuming that the multiplex load (q) is 17. This indicates that transmission may be performed using a modulation depth that takes into account multiple load gain. Although ideal conditions have been set above, consider the actual system operating conditions. In this case, the positions of the mobile radios 100 are randomly scattered and the radio wave propagation pattern [j,1] changes variously, so the reception power of the radio base 7430 will vary for each time slot. In addition, each mobile radio device 10
The carrier waves from 0 are also not necessarily phase synchronized. Therefore, if the depth of modulation is large in a time slot where the reception level is high, it is expected that the adjacent time slots will be adversely affected due to the influence of multiplex radio wave propagation. However, this can be alleviated by taking measures such as increasing guard time. In addition, in the case of the small zone method, as co-channel interference,
There is no need to worry too much about the possibility that the transmitted waves of one mobile radio device 100 may cause interference to other radio base stations 30 that are located at different locations; on the contrary, there is a possibility that the transmission waves can be used to repeatedly reduce the number of zones. be. It uses multiple load gain as FM (PM) modulation, and as a result of deep modulation,
This is because 1q of broadband bands can be reduced by 1q, and resistance to co-channel interference is increased. Taking all the above into account, the system is designed on the assumption that when the mobile radio 1100 transmits and when the radio base station 30 receives, substantially the same multiple load gain can be obtained as when transmitting from the radio base station 30. It became clear that it is possible. (5) Specific method of utilizing the multiple load gain The multiple load gain in the case where one frame length is 1 m5ec and the number of multiplexes of TCM (time division time compression multiplexing) is 500 is determined, and an example of its use will be explained. First, S CP C (Single Channel
Per Carrier. Find the signal S to noise N ratio of analog FM (in which the signal of one telephone channel is modulated on one carrier wave), and calculate this and TC
Compare with M. The level (voltage value) of the input signal to the receiver is C, the FM modulation index is mf, and the noise level (voltage value) per unit frequency is n. Assuming that the bandwidth of the low frequency amplifier in the discriminator output stage is Fa, and that the highest modulation frequency fa of the modulated wave is equal to Fa, the signal-to-noise ratio is given by the above equation. 1/2 -1/n S/N=3 m(C(2Fa) (24) This formula is quoted from the following literature. Edited by Sugawara "'FM Radio Radio Engineering Usukan Kogyo Shinbunsha TI Showa 3
4th year page 401 (13, 25) Formula Next, the TCM signal with multiplexing number Q is FM under the following conditions.
The S/N in this case is given by the following formula. (S/N>, C, - 31z2mfQcQ (2FaQ) -172(25) However, F8o = QF8 m r o = Q m f no = Qn (25') In other words, in the TCM signal, the frequency of the original signal is multiplied by Q. For the sake of
The bandwidth of the low frequency amplifier has increased by a factor of Q, and the depth of modulation (modulation index) has also increased by a factor of a factor of Q. Therefore, the bandwidth of the 1r level has also increased by a factor of a factor of Next, the received signal level C6 of the TCM so that the S/N on the left sides of equations (23> and (24) take the same value) is determined by the following equation: 3172m, oCo (2Eao) -172 (26) From the formula, 1/2 (27) C, = Q C is 1q.In other words, the reception level is 01 in voltage.
72 times, which means that Q times the power level is required. Therefore, the transmission power is 01 from 5CPC.
Is it necessary to increase it by 8? Next, find the multiple load number 1q of the TCM signal in the above example. As explained in section (3) above, in this case, FDM
The equivalent multiplex number is n'-500xl/6000 (sec> ÷ (1/1
000 (5ec) ) = 500xl/6 = 83
CH Therefore, from FIG. 8, it can be seen that the multiple load gain is an intermediate value between 28.6 dB for 60 channel multiplexing and 32.6 dB for 120 channel multiplexing. By creating the graph shown in Figure 7 and estimating it, we obtain a multiple load gain of 30 dB.In addition, the modulation depth (
To deepen the shift> and gradually reduce the transmission power, use the multiple load gain here. TCML. Assuming that the transmission output for a 1-channel analog FM signal is 10mW, which is at the cordless telephone level, the required transmission power in this case is 50mW using equation (27).
After multiplying by 0, subtract (by multiple load) to obtain (10mWx 500) -30dB=10mWx
We get 500/1000=5.0mW (28). In other words, using TCM requires less power. Next, we will explain the physical meaning of (by multiple load) q in the CM signal, and discuss points to be noted when using this as a system. CM signals have a long frame period of 1 second or more (the number of multiplexes is 6).
000), the multiplex load 1q cannot be obtained at all, but in this case, the FM modulation index of the TCM signal has a constant value determined by the system. For example, the modulation index of the original signal (0.3~3.0KH2) is 1.75K! -1z (1KH2 tone signal with standard modulation deviation), and in the case of TCM that multiplexes 500 of these, the signal band is 150 to 1500KH
z, the standard modulation deviation is 875 K I-I Z (
(When a 500KH2 tone signal is standard modulated). However, if the frame length is set to 1 m5ec, 30 dB can be obtained for each multiple load (q) as shown in the above example, and this 1 q for each multiple load is used to increase the depth of modulation, but the actual modulated wave Let me explain how it works. First, let's consider an all-channel implementation, that is, a case where a telephone signal is flowing in all time slots. In this case, the effect of a multiple load gain of 30 dB on modulation deviation increase is , Considering from Fig. 9, the multiple load gain is the relative power of the sine waves with the same peak value, which is approximately +8 dB. It can be seen that it is equal to the modulation shift of the signal when only slots are used.Next, consider the case where the load is gradually increased from implementing all time slots. The problem is what happens to the modulation deviation of the signal when some of the signals are used for actual sound transmission, and the others are not used as empty time points.In this case, the implementation Since the number of channels decreases, the multiple load gain naturally decreases.For example, 1/2 of 25
In the case of 0￥菰, the multiple load gain is n' = 250xl/6000÷(1/1000)- from equation (23).
250X1/6 = 42 (CH) Therefore, from FIG. 7, it can be seen that the multiple load gain 1 is 24.5 dB. However, since the load is 172, the power level of the modulated signal is lowered by 3 dB. Therefore, in this case, the geometric multiple load function) q becomes 27.5 dB,
Although the effective modulation depth of Ding CM-FM is a little off, it is considered that this has no effect on system operation. Furthermore, the effective modulation shift is determined when the number of implementations is reduced and only one time slot is used. The multiple load gain of 1 channel is 0 (1B, or the signal load is 11500 compared to the full implementation, which is a decrease of 27 dB. Therefore, the apparent multiple load gain is 27 dB.
Therefore, even if the modulator is operated with this value set to 30 dB, it may be assumed that the system operation is not affected. In addition, in the manual stage of the modulation circuit of an actual radio, (DC(In5t
Antaneous Deviation Contr
A circuit (instantaneous modulation deviation amount suppression) is provided, and is provided with a function of limiting the modulation depth to a certain value or less. Therefore, as a modulator output, the effective modulation deviation is kept below a certain value, regardless of the implementation state of the CM telephone signal. As explained above, multiple fQ of the TCM signal
(It has become clear that by using the 7i gain to increase the modulation deviation of the M signal, it is possible to significantly reduce the transmission output. Technically speaking, this has a very large effect on power saving. In other words, if a 10mW wireless device for continuous transmission at 5cpc is operated at a time rate of 11500, or 0.2%, its output is only 5mW of 10mW, which is 172%, so there is a power saving effect. It is obvious that 1 is large. The advantages and disadvantages of an example in which a guard time is set between the time slots of the TCM signal according to the present invention will be explained. It is not always necessary to provide a guard time between pulse trains as in the case of zero-bow. However, it is necessary to eliminate the effects of timing shifts in synchronization signals and delay waves caused by multiple waves on radio wave propagation. Therefore, a guard time B2 may be added between time slots.The specific number 1a of the guard time varies depending on the applied Liebe system, but for example, for an indoor mobile phone system, it is O11 to 0.5 μSec, Approximately 5 to 10 μsec is appropriate for automatic telephone calls.In a system with a guard time, if the frame length is constant, the time width of the slot time decreases by the guard time, so the original signal must have a high compression ratio, and therefore the highest frequency of the signal will be high.In the cordless telephone example above, time slot 1~ is 1 m5ec: 500 = 2 μsec;
If a guard time of −10%, that is, 0.2 μsec is taken, the time slot becomes 1.8 μsec. Also, the highest frequency is 3 (Z) without guard time.
x 500 = 1.5MHz, taking 10% guard time, 1.5Mt1z xlO/9 = 1.
67 MHz. Therefore, the required bandwidth becomes correspondingly wider, and the frequency effective utilization rate decreases by 11%. Next, we apply multiload gain to the amplifier design. In this case, the level of the multiplexed voice converted into TCM may be considered to be lower than the level conventionally considered by the multiplex load gain jq. Therefore, even if the amplification factor can be increased accordingly, or the output level can be made higher than the conventional one by the amount of the multiple load gain, the distortion factor and the like will remain at the values conventionally assumed. Multiload gain is applied not only to active circuits as described above, but also to
It is also applicable to passive circuits as described below. That is, if applied to a microcircuit, the rated output can be increased in level by the amount of the multiple load gain, and it can be operated in the operating state previously assumed. This is wireless transmission (1
If applied to four aircraft, the following benefits will be obtained. For example, inserting a power amplifier into the output of the transmission mixer 133 in FIG. 1B-1 is often used to increase the range of radio waves. In this case, by introducing the multiple load gain 14, it is possible to make the transmission output level higher than the conventionally assumed level by the amount indicated by the multiple load gain. Alternatively, if the same transmission level as the conventional one is sufficient, the rated output of the amplifier can be made with a lower level output than the conventional one by the amount of the multiple load gain. The above concept of rated power can be applied not only to transmission mixers but also to all resistors, capacitors, inductances, etc. (6) Example of a system in which the present invention is used to transmit composite signals such as voice, data, images, etc. What was explained in sections (3) and (4) above is a system in which the present invention is applied to voice signals, and・In this case, the signal accommodated in a slot is only one audio channel.However, in the present invention, such a constraint is not necessarily necessary, and a wide variety of composite signals as described below are used. First, although the signals used for communication include low-speed data signals,
A system in which the signal is mainly an audio signal and the number of audio channels accommodated in one time slot is two will be described. This is l5DN (Integrated Service Digital Network)
It provides communication quality that is compatible with the times, and is equivalent to 28+D digital communication (2 audio channels + 1 digital channel). FIG. 3C is a spectrum diagram when two audio channels and one data channel are accommodated. In the figure, control signals and communication signals CH11°CH21...
, CHrll have the same frequency band as the control signal and speech signals C, H1, CH2, . . . , CHn in FIG. 3B. On the other hand, the call signals CH12, CH2
2,・---・CHn2 are Cl-11, C, respectively
H2 and CHn are shifted to a higher frequency band by 45 KHZ on the frequency axis, and this can be realized by a method similar to that used when multiplexing known FDM (frequency division multiplex) signals. Further, in FIG. 1C-1, all input/output signals to the signal processing section 31 must be sent and received as analog multiplexed signals for two channels of audio signals and one channel of data signals. If the 15DN interface is 28+D, it is necessary to provide an analog-to-digital conversion circuit within the signal processing section 31. FIG. 1D-1 shows an example of the signal processing section 31 that satisfies such requirements, and the human input signal from the signal speed restoration circuit group 3B is processed by an A/D converter 82,
An output to the signal speed conversion circuit group 51 is applied via a D/A converter 83. Similarly, data signals CH13, CH
23,------, CHn3 has also changed frequency to 1q
It's something that was given to me. These three signals are accommodated within the same time slot. For example, the SDI accommodates speech signals CH11, CH12, and data signals CH13. It is clear from the explanation in section (3) that the multiple load gain of the composite signal of voices 1q as described above is larger than that in the case where one voice is accommodated. . That is, it can be applied by substituting 2n in place of n on the right side of equation (21). However, the existence of a data signal was negligible due to the narrow bandwidth. When building an actual system, 1 time
The only difference is that the highest frequency of the signal within the slot increases by the addition of the data signal, and there is no change in the calculation of the multiple load gain. Next, a composite signal contained in one frame of an audio and image signal or a TCM (time division time compression multiplex) signal will be explained. However, both audio and images are analog format 0.
Wings. Regarding the multiple load gain 17 that occurs when a large number of image signals are multiplexed, the clear value is not known, but for example, when transmitting an image signal showing a person, the information of each frame screen changes. It can be easily estimated that if a method such as transmitting only the minutes is used, the multi-handed load gain will be reduced by j7. Therefore, the number of multiplexed image signals is p, the highest frequency is rh, and the multiple load gain of 17 at this time is q. On the other hand, if we assume that the time of one frame is the same as above when transmitting the above image signal as a TCM signal, the multiple load gain when each image signal addressed to one image signal is accommodated in all time slots is: D' -D/ (2rhd>(21'
) is equal to the multiplexing gain q' of a frequency division multiplexed image signal having the multiplexing number p'. Now, when accommodating n audio channels and m video channels in one frame, the most efficient method for frequency utilization is to make the multiple load gains of both signals equal and the highest frequencies of the signals equal. The good news will be clear from the explanation already given. Next, an example of a system that transmits and receives audio (telephone) and images will be explained using FIG. 2C. accommodating, wireless base station 30B
The configuration of the time slot is shown when the mobile radio 100B transmits and the mobile radio 100B receives. Here, the number of time slots included in one frame is 30. This is an example in which the highest frequency of the image signal is 10 times that of voice (telephone), and the multiple load gain of the voice (telephone) 30 channels is equal to the multiple load gain of 30 image channels. Under these conditions, If the frame structure shown in Fig. 2C (a) is adopted, the effective frequency utilization will be the best compared to the frame structure of other TCM signals that transmit the same female transmission signal.In addition, Fig. 2C (a) Although a guard time is provided between the time slots of , for example, between SDl and SD2, this is not always necessary;
As shown in FIG. 2A, the guard time may be configured to be substantially zero. However, when used for outdoor mobile communications that travel at relatively high speeds, such as in automobiles, it is better to provide a card time because it is affected by multiple wave propagation. The example of the composite signal shown in FIG. 2C (b) is a case where the multiload gain of 1.00 image signals is equal to the multiload gain of 200 voice (telephone) signals. However, These values are equivalent to the number of multiplexed FDM signals.In this case, since the number of multiplexed communications is different for image and voice (telephone), multiplexing within one time slot as shown in (a) of the same figure is required. However, as shown in FIG. 2C (b), the highest frequencies of the image signal and the audio (telephone) signal can be accommodated in each time slot within one frame, and the highest frequencies of the image signal and the audio (telephone) signal can be respectively , if the time length of one frame is the king, the degree of compression of each signal is ψ9.ψ,
is given by the following formula. Signal compression degree φ of image signal, = T/(2fvh) (29a) Audio (
Telephone) signal compression degree ψ, =T/ (2f, h) < 29 b)
Therefore, the highest frequency F, h within a time slot of each image and audio (telephone) signal. Fph is ``Vh=ψvfVh (30a)Fph
-ψ, f, h (30b) To obtain a TCM signal with high frequency efficiency, (30a>, (30b)
In the formula, it is desirable that Fvh=Fph (31). Therefore, ψvfvh−ψ, −f, h (32) (32)
The conditions of equation (29) are satisfied as long as equation (29) is satisfied. In such a system, the telephone section 101 of the mobile radio device 100 shown in FIG. 1B-1 naturally transmits image signals and audio (
Since it will handle both telephone (telephone) signals, it will be called the image/telephone section. 1B-2
The configuration is as shown in the figure, and its configuration can be described as a commonly used wireless videophone terminal. A wireless videophone as explained above *At the end, 1B
- The same system as the telephone terminal shown in Figure 1 -
This will be explained using FIG. 2, FIG. 1D-2, and FIG. 1D3. In FIG. 1D-2, a voice (telephone) image/telephone 1 is input from one side of the signal processing section 31. The signal processing unit 31 identifies these signals, sends the voice (telephone) signal to switch STA as shown in FIG. 1D-2, and the image signal (also to switch 8B). ) An identification signal is included in the signal so that it can be distinguished whether it is a single signal or a composite signal of audio (telephone) and images.The subsequent operation is as already explained, but the composite signal This is transmitted to the mobile radio device 100B that can receive this, and the voice (telephone) single signal is transmitted to both the mobile radio devices 100 and 1008 in FIG. 1B-1 or FIG. 1B-2. This is because the mobile radio 100B may want to receive only voice (telephone) signals. Under the design conditions described above, the number of FDM conversion equivalent time slots in one frame is 100 for video and 100 for telephone. 2
00, a total of 300. Depending on the system, the number of time slots may not be too sensitive, and this will be explained below. Image 1100f Yarnell and telephone 200-) It is assumed that the multiload gains of Yarnel are equal as described above, and time slot 1
~I want to set the number to 200, including 100 images and 100 phone calls. This can be achieved by multiplexing two telephone channels within the telephone subframe in an FDM manner, as shown in FIG. 2C (C). In addition, when the number of time slots in one frame is reduced by multiplexing telephone signals in this way,
Compared to the example system described above, one frame of B
), the multiple load gain will increase. Therefore, when it is not necessary to increase the multiple load gain, it is possible to increase the time length of one frame accordingly. [Effect of the invention 1] As is clear from the above explanation, the multiplex gain of the time-division time compression multiplexed signal, which has not been clearly shown in the past, has been quantitatively clarified using system parameters. For example, even if the angle modulation depth (deviation) is increased by the amount of the heavy load gain, the influence on other wireless channels can be kept within the conventional design value, and the wireless It becomes possible to gradually reduce the transmission output level per channel compared to conventional systems, resulting in power savings. In general, as we enter the 15DN era, it is possible to construct systems capable of two telephone channels, one data channel, or simultaneous transmission of telephone calls and images, which has an extremely large effect on wireless communication systems.

[Brief explanation of drawings]

FIG. 1A is a conceptual configuration diagram showing the concept of the system of the present invention, FIG. 1B-1 and FIG. 1B-2 are circuit configuration diagrams of a mobile radio device used in the system of the present invention, and FIG. 1C-1. 1C-2 and 1C-2 are circuit configuration diagrams of a radio base station used in the system of the present invention, respectively. Detailed circuit configuration diagram; Figure 2A is a time slot structure diagram for explaining the time slots used in the system of the present invention; Figure 2B is a diagram showing the radio signal waveform of the time slot; Figure 2C is a diagram showing the radio signal waveform of the time slot. is a time slot MRM diagram when transmitting and receiving telephone signals and image signals; Figures 3A, 3B, and 3C are spectrum diagrams showing the spectrum of telephone signals and control signals; 4A and 4B are flow charts showing the flow of the calling operation of the system according to the present invention; FIG. 5 is a spectrum diagram of a frequency division multiplexed signal; FIG. 6A is an amplitude diagram showing changes in the amplitude of the time-division time compression multiplex signal; FIGS. 6B, 6C, and 6D are sampling diagrams showing how the time-division time compression multiplex signal is sampled; Figure 7 is a diagram showing the relationship between the multiple load gain of a time division time compression multiplexed signal and the number of multiplexed audio signals, and Figures 8 and 9 are diagrams showing the relationship between the multiple load gain of a frequency division multiplexed signal and the number of voice signals cited from known literature. FIG. 17 is a diagram showing the relationship with the number of routes. 10...Telephone m 20...Gateway exchange 2
2-] to 22-n...Communication signal 30...Radio base station 31...Signal processing section 32
...Wireless transmission circuit 35...Wireless reception circuit 38.
...Signal speed restoration circuit group 38-1 to 38-n...Transmission speed restoration circuit 39...
Signal selection circuit group 39-1 to 39-n...Signal selection circuit 40...Control unit 41...Clock generator 42...Timing generation circuit 51...Signal speed conversion circuit group 51-1... 51-n...Signal speed conversion circuit 52...
Signal assignment circuit group 52-1 to 52-n...Signal assignment circuit 87...4-wire to 2-wire converter 88...A/D converter 89...D/A converter 9
1...Digital encoding circuit 92...Multiple conversion circuit 00.100-1 to 100-n-...Mobile radio device 01.
...Telephone section 102...Image terminal section 20...
Reference crystal oscillator 21-1.121-2...Synthesizer 22-1.1
22-2...Switch 23...Intermittent transmission/reception i! , 11 Submersible unit 31...Speed conversion circuit 32...Wireless transmission circuit 34...Transmission section 36...Reception mixer 3B...Speed restoration circuit 133...Transmission mixer 135...Wireless reception circuit 137... Receiving unit 141... Clock regenerator.

Claims (1)

[Claims] 1. Each radio base means (30) each covering a plurality of zones to constitute a service area, and a radio base means (30) for moving across the plurality of zones and communicating with the radio base means; barrier exchange means (20) for exchanging communications with each mobile radio means (100) using a radio channel carrying time-compressed delimited signals in time slots of a frame structure; In mobile communication methods,
The time-compressed segmented signal may also include a frequency division multiplexed signal of a plurality of signals, and the radio base means and the mobile radio means are separated based on the multiple load gain obtained by the segmented signal. A time division communication method for mobile communication in which a degree of multiplicity and a degree of compression within the time slot to be used for communication during the time slot are determined. 2. Each radio base means (30) each covering a plurality of zones to constitute a service area, and a time slot arranged in a frame for moving across said plurality of zones and communicating with said radio base means. In a mobile communication system using a barrier exchange means (20) for exchanging communication between each mobile radio means (100) using a radio channel carrying time-compressed segmented signals, The time-compressed segmented signal may also include a frequency division multiplexed signal of a plurality of signals, and the radio base means and the mobile radio means are divided based on the multiple load gain obtained by the segmented signal. A time division communication system for mobile communication in which a degree of multiplicity and a degree of compression within the time slot to be used are determined.