JPH03255735A - Method and system for time division communication of mobile object communication - Google Patents

Abstract

PURPOSE:To improve profitability by inserting a compressor to the input of a transmitter, changing the input to desired distribution and executing demodulation by providing an expander at the side of a receiver. CONSTITUTION:Mobile object communication is executed by using respective radio base means 30 to respectively cover plural zones and to constitute service areas, and a gate exchanging means 20 to exchange communication with respective mobile radio means (100-1)-(100-n) which are moved while crossing the plural zones and use radio channels loading signals hourly compressed and divided to time slots for the communication with the radio base means 30. In such a case, at first, the level of a radio signal to be used for the communication between the radio base means 30 and the mobile radio means (100-1)-(100-n) based on multiple load gain to be obtained by the hourly compressed and divided signal is determined. Next, the signal to be transmitted is transmitted by compressing the amplitude distribution to the desired range and the received signal is restored to the original distribution.

Description

DETAILED DESCRIPTION OF THE INVENTION 1. Field of Industrial Application] The present invention relates to a time division communication method and system for a wireless communication channel in mobile communication, and to an effective use of multiple load gain of a time compression multiplex signal, which is a modulation signal. . More specifically, a certain radio channel is given, and a radio channel is being set up and communicated with one of the many mobile radios in the service area or with an opposing radio base station using this channel. When another mobile radio starts communicating with another radio base station using the same radio channel, due to the effective use of frequencies or radio wave propagation characteristics, the communication between the mobile radio and the radio A method for eliminating adverse effects on communication with a base station, improving power saving of a transmitter by reducing transmission output, and improving effective frequency utilization, and an economical method using the method. The aim is to provide a system that is easy to use.

2. Prior Art: In mobile communication using voice using the small zone method, a method employing time division time compression multiplexed signals is described in the following literature.

Literature 1. Ito “Study of mobile phone system - Time division time compression F
~Proposal of 1 modulation method-°' IEICE technical report RC389-
11 July 1989 literature 2. Ito “Study of mobile phone system - Time division time compression F
Theoretical study of M modulation system Technical Report RC389-39, IEICE
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. Built-in in mobile radios and radio base stations for high-speed readout in predetermined time slots, angle modulation or width modulation of carrier waves with signals accommodated by these time slots, and time-intermittent transmission and reception. a radio receiving circuit having receiving mixers facing each other and communicating with each other, a radio 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. A switch circuit is provided for each to intermittent the output of the synthesizer applied to each, and this intermittent state is synchronized for both transmission and reception, and the same intermittent transmission and reception as above is synchronized for the radio base station communicating with the mobile radio. In order to extract only the signal accommodated in the predetermined time slot, the receiving side opens and closes the radio receiving circuit to receive the signal, demodulates the signal, and stores the obtained signal in the storage circuit. 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 it out at a low speed that is 1/n of the speed at which it is stored in this memory circuit. ing.

In addition, Reference 2 examines adjacent channel interference and co-channel interference, which are problems when applying the TCM (time division time compression multiplexing)-FM method described above to small zones. Possibilities of realizing the system have been shown by appropriately selecting 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 Theory for
r Hultichannel Amplifiers
8S block J 18 oct, 1939 document 4
．． C, B, Feldman et al.”Band Width
and mission performance
ance” BSTJ, July 194
Figure 11, pages 9490-595, was created from Fi (7, 7) of the above-mentioned document 3, and Figure 12 was quoted from page 495 of the above-mentioned document 4. This shows that it is possible to obtain substantially the same multi-load gain as in the previous example.

The reason why multiple load gain can be obtained and its application to wireless angle modulation will be briefly explained below.

The level of a telephone channel in operation, through which telephone signals are flowing, varies from person to person, gender, and subscriber line length; Be sure to keep an interval. Also, while the other party is talking, the other party does not speak and the other party does not add any signals. Do not speak during exchange connection. For this reason, the individual signal levels are 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 (33: single 5
The level of the method using 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 is small, the influence of each voice on the synthesized signal, which fluctuates greatly, is not direct. However, 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, )t.

Brook and J. T. Dixon (cited above, reference 3) were statistically determined for telephones in the United States. According to the results, the change in power of a sine 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 the multiple telephone signal is, the multiple load gain shown in FIG. 11 is compared with the sum of the peak voltages of the individual signals. That is, for example, in the 960 channel system, six channels are loaded at the maximum at the same time, resulting in the same peak voltage as if the signal of the 954 channel was not loaded.

In the SS-FM system, fluctuations in the voltage of the composite signal result in frequency deviations, so when the peak frequency deviation of the composite signal is set to a certain value and the number of multiplexed communication channels N increases, the voltages of each communication signal are summed. Compared to the case in which the modulation index per channel can be increased by the multiple load gain shown in FIG.
S/N given when starting frequency deviation is set to an arbitrary value
It will be improved that much more.

[Problem to be solved by the invention]

In the system construction examples shown in the above-mentioned documents 1 and 2,
It does not disclose the existence of a multiple load gain in time-division time compression multiplexed signals transmitted from a radio base station to a large number of mobile radios, and does not utilize this multiple load gain.

Therefore, if this multiload gain analysis had been done, there are many benefits that could be gained in system design, such as the reduction in transmitted power level possible by increasing the depth of frequency modulation. Ya, T.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. I felt that this was an issue that needed to be solved.

What was disclosed in References 3 and 4 clarified the multiple load gain in so-called frequency division multiplexed signals, which are obtained by converting the frequency of audio signals and multiplexing them so that they do not overlap on the frequency axis. It cannot be applied to time compression multiplexed (TCM) signals, and the existence of multiple load gain is unknown. (References 1 and 2 above)
There was an issue to be solved in that it was not possible to concretely realize the same thing as in the case of

[Means for solving the problem: The number of multiplexed TCM (time division time compression multiplexed) signals (number of communication channels), the time length of one frame, and the highest frequency of the original signal are taken as parameters, and the multiplexed nine load of the TCM signal is calculated. Using the sampling theorem, the gain was clearly derived in relation to the multiple load gain in an FDM (frequency division multiplexed) signal, and this was made practical.

Roughly speaking, a compressor (
A compressor was inserted to change the distribution to the desired distribution, and an expander was installed on the receiver side for demodulation.

[Effect] Since it has become clear that there is a multiple load gain even in the CM signal, it is now possible to specifically calculate the multiple load gain using various design parameters of the system, and it is possible to tolerate interference, etc. FM (PM) while keeping it within the field.
By deepening the degree of modulation, it is possible to gradually reduce the transmission output. Since the peak value of the output signal from the transmitter no longer increases beyond a certain value in modulation deviation, it is possible to eliminate system operational obstacles, and as a result, the modulator and It has become easier to design amplifiers, 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, and FIG. 1C 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 exchange 112
0, a circuit to convert signal speed, a circuit to allocate and select time slots, and a control section.
100 (100-1 to 100-n>) and has a wireless transmitting/receiving circuit that transmits and receives wireless signals.

Here, between the barrier switch 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 CH1 to CHn and control signals.

FIG. M1B shows the circuit configuration of the mobile radio device 100 that communicates with the radio base 830.

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 is sent to a speed restoration circuit 138 and a control section 140. is input to the clock regenerator 141. The clock regenerator 141 regenerates a clock from the received signal and sends it to the speed recovery circuit 13.
8, the control section 140, and the timing generator 142.

The speed recovery circuit 138 calculates the speed (pitch in the case of an analog signal) of the compressed and segmented communication signal in the received signal.
is restored and input as a continuous signal to telephone section 101 and control section 140.

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 to send a radio signal from the mobile radio device 100 to the radio base station 30 using the time slot that is permitted to be used, timing information from the timing generator 142 shown in FIG. 1B is sent to the control unit 140. It is necessary that it has been obtained through

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. The speed conversion circuit 131 supplies necessary timing to the speed restoration circuit 138.

This mobile radio device 100 further includes a synthesizer 121.
-1iB and 121-2, and the changeover switch 122-1.
122-2 and selector switches 122-1, 122'-2
'-transmission/reception intermittent controller 123 that generates signals for switching respectively. ! Synthesizers 121 - 1 and 121 - 2 , transmission/reception intermittent controller 123 , and timing generator 142 are controlled by control section 140 . Each synthesizer 12
1-1.12'1-2 is supplied with a reference frequency from a reference crystal oscillator 120.

A wireless base station 30 is shown in FIG. 1C.

N-channel communication signal 22-1 with barrier exchange R20
22-n are connected to three signal processing units forming an interface through transmission lines.

Now, the communication signal 22- sent from the opening switch 20
1 to 22-n are input to the signal processing unit 31 of the wireless base station 30. If the signal processing section 31 is equipped with an amplifier to compensate for transmission loss, so-called 2-wire to 4-wire conversion can be performed. That is, the input signal and the output signal are mixed and separated, and the input signal from the barrier exchange 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, the input signal from the barrier exchange 1120 is input to the signal speed conversion circuit group 51 including many signal speed conversion circuits 51-1 to 51-n, and is subjected to speed (pitch) conversion at predetermined time intervals. .

Furthermore, the signal transmitted from the radio base station 30 to the gateway exchange 20 is determined by the output of the radio receiving circuit 35 being transmitted to the signal selection circuit group 39.
is input to the signal speed restoration circuit group 38 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 139,
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 sent to a signal processing section 31. After being input and subjected to 4-wire to 2-wire conversion, this output is sent to the barrier switch 20 as communication signals 22-1 to 22-2.
2- Sent as self.

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.
argeCoupled Device), B
BD (Bucket Br1(fadeDevice
), and the memory used in television receivers and tape recorders that compress or expand the time axis of conversation can be used (References: Koita et al. °' Time axis of conversation (Nikkei Electronics, July 26, 1976, pp. 92-133).

The circuit using the CCD or 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 document. 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 signal output from the barrier switch 20 via the signal processing section 31 is input to the signal speed conversion circuit 051, and after speed (pitch) conversion processing is performed, a signal is generated to allocate the signal to each time slot. It is applied to the assignment 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 communication signals are viewed in terms of channels, they are arranged in series without overlapping in chronological order, and when all control signals or communication signals (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), 5D2
−． 5Drl is characterized by the fact that speed-converted communication signals are allocated to each time slot.

Here, as shown in the figure, one time slot accommodates a synchronization signal, a control signal, or a call signal. 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 32
In this case, 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 (250 Hz>) or the high frequency side (3850 Hz>) can be used. This signal is used, for example, when it is desired to send a control signal during a call.

FIG. 3A (b) shows an example of an in-band signal, which is used when making and receiving calls.

The above examples were all tone signals, but by increasing the number of tone signals or modulating the tone to create a subcarrier signal, it is possible to transmit many 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, it is also possible to digitally encode the audio signal and time-division multiplex the two to transmit. The circuit configuration of is shown in FIG. 3C. FIG. 3C shows an audio signal digital encoding circuit 9.
This is an example of a case where the data signal is digitized, multiplex-converted with the data signal by the multiplex conversion circuit 92, and applied to the modulation circuit included in the wireless transmission circuit 32. However, in the case of digital data signals, there is usually no multiple load advantage when multiplexing analog signals, which will be described later, so this point must be taken into consideration when designing the system.

Then, it is received by the opposite receiver, and the 3rd C
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 a mobile radio device 100-1, 100-2°...,
100-n shows a transmission signal addressed to the wireless base station 30.

Also, each time slot 5LJ1. SU2,...,s
If the contents of un are shown in detail, as shown in the lower left of FIG.

However, depending on the short distance between the radio base station 30 and the mobile radio device 100 or the signal speed, it is possible to omit the synchronization signal. 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 size 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 4o, 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), thereby faithfully reproducing the original signal and transmitting it to the gateway exchange 1120.

The mode of transmission in the signal space provided by the present invention will be described below using the required transmission band and the relationship between this and adjacent wireless channels.

As shown in FIG. 1C, the control signal from the control section 40 is applied to the wireless transmission circuit 32 in parallel with the output of the signal allocation circuit group 52. However, depending on the size of the speed conversion rate, it is also possible to apply the signal to the wireless transmission circuit 32 from the output of the signal allocation circuit group 52 after performing the same processing as the call signal.

Next, as shown in FIG. 1B, the mobile radio device 100 also has a circuit configuration required when the radio base station 30 has one communication channel among its functions.

Original signal, for example, an audio signal (0.3KHz to 3.0KH2
The frequency distribution on the output side when the signal passes through the signal speed conversion circuit group 51 (FIG. 1C) is shown in FIG. 3B. That is, if the audio signal is converted 15 times as described above, the frequency distribution of the signal is as shown in Figure 3B < 4
．． This means that it has been expanded to 5KHz to 45KH2. Although the frequency distribution of the signal has been expanded here, it is necessary to keep in mind 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. It is assumed that among these signals, a control signal (0, 2 to LOKH2) and a call signal gcH1 (4, 5 to 45 KH2, expressed as SDl) are accommodated in a time slot, for example, SDl.

The same applies to the audio signals accommodated in the other time slots SD2 to SDn.

That is, time slot sDi <1=2°3, -
, n> accommodates a control signal <0.2 to 4.0KH2) and a communication signal C1-1i (L5 to 45KH2). However, the signals in each time slot are arranged in chronological order, and signals in multiple time slots are not simultaneously applied to the radio transmitting circuit 32.

The wireless transmission circuit 32 along with these call signals or control signals
When added to the angle modulator included in the angular modulator, the required transmission band requires at least fo±45 KHz. 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
,. , and let fH be the maximum signal speed due to the speeding up of the audio signal of j-ho above, 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. It is sent to the barrier exchange 20.

FIGS. 1D and 1E show another embodiment 100B of the mobile radio 100 and a baby 100C. The difference from the mobile radio device 100 shown in FIG. 1B is that a compression operation is performed to prevent the peak value or modulation deviation of the output signal from the transmitter from increasing beyond a certain value. The main difference is that it is equipped with a compressor 171 to compress the signal, and an expander 172 to expand the signal compressed by the compressor.

In the mobile radio device 100B (FIG. 1D), the compressor 171 includes a speed conversion circuit 131 and a wireless transmission circuit 132.
The expander 172 is provided between the radio reception circuit 135 and the speed restoration circuit 138.

In the mobile radio 100G (FIG. 1E), the compressor 171 is provided between the telephone section 101 and the speed conversion circuit 131, and the expander 172 is provided between the speed restoration circuit 13.
8 and a telephone unit 101 is shown.

Unless otherwise specified below, mobile radio equipment 100.10
08.100C are collectively referred to simply as the mobile radio device 100.

Other embodiments 30 and 30C of the radio base station 30 are shown in FIGS. 1F and 1G. The difference from the radio base station 30 shown in FIG. 1C is that the compression operation is performed to prevent the peak value of the output signal from the transmitter from increasing beyond a certain value in terms of modulation deviation. Compressor 71
Alternatively, a compressor group 71 including compressors 71-1 to 71-n and an expander 72 for conversely expanding a signal compressed by the compressor or an expander including expanders 72-1 to 72-n. The point is that the group 72 is provided.

In the wireless base station 30B (FIG. 1F), the compressor 71 is provided between the signal allocation circuit group 52 and the focal line transmitting circuit 32, and the expander 72 is connected to the wireless receiving circuit 35 and three selection circuit groups. is set in the darkness. In the wireless base station 300 (Fig. 1G) h-:B-', the compressor 71-1""-7'l n @contains L compressor group 7'1 is 1 word compressor 31 and 1 word Bow speed conversion circuit details 5
1, and the expanders 72-1 to 72-n
An example is shown in which the expander group 72 including the signal speed i is connected to the circuit details 38 and the Shinzuki sakai part 31.

Below, and <1. If zfn does not change, the wireless base station 30
．． 308.300 are collectively referred to as the M-broken base station 30.

Next, the call originating/receiving operation of the system according to the present invention will be explained for the case of voice i number 854.

(-) Call origination from the mobile radio device 100 will be explained using the flowcharts shown in FIGS. 4A and 4B.

9-motion radio! ! When i100's haze source is turned on,
The Retsu 9 receiving circuit 135 in FIG. 1B starts capturing the control signal included in the downlink (radio base station 30 - mobile line equipment 100) radio channel (channel CH1). If the system is provided with multiple radio channels, the radio channel with the highest received input electric field; the radio channel indicated by the control signal contained in the radio channel; the time slot within the radio channel; Among them, channels with empty time slots and other wireless channels (hereinafter referred to as channel C-11) that follow the procedures determined by the system enter the receiving state. This is possible by capturing the synchronization signal within the time slot SDi shown in FIG.
A control signal is sent to the synthesizer 1-1 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 is off-hook (starts making a call) (S201, FIG. 4A), the synthesizer 121-2 of FIG. 1B generates a local frequency that enables transmission of the wireless channel CH1. A control signal is received from the control section 140 to cause the control to occur.

Further, the switch 122-2 is also turned toward the synthesizer 121-2 side and becomes fixed. Next, wireless channel CH1
The call control signal outputted from the telephone unit 101 is sent using the telephone unit 101. This control signal has a frequency band shown in FIG.
It is sent using un.

The transmission of this control signal is limited to the time slot Sun, and is sent in bursts, and no signal is transmitted during the Sun time slot, so it does not adversely affect other communications. However, if the speed of the control signal is relatively slow or the amount of information in the signal is too large to be accommodated in one time slot, the same time slot of the next frame will be sent after one frame or roughly. - Transmitted using slots.

Specifically, the following method is used to capture the time slot Sun. The control signal transmitted from the radio base station 30 includes a synchronization signal and a subsequent control signal, as shown in FIG. It becomes possible.

Furthermore, this control signal includes control information such as currently used time slots and unused time slots (empty time slot display).

Some systems use time slots 5D (i=1.
2. ..., n) or other communications, it may contain only synchronization signals and speech signals; By receiving this control signal, the mobile radio 100 can know which time slot to use to send out the calling signal.

Hey, 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 other systems, empty slots may not be displayed within the time slots of , and in this case, the following audio multiplex signals SD1, SD2, . ..., S
Is it necessary to search for the presence of Dn one after another and check for empty time slots?

Now, in the main discussion, the mobile radio 11100 that has received the control information sent from the radio base station 30 by one of the above methods should send out a call control signal in which time slot. It is possible to make a judgment including the transmission timing.

Therefore, assuming that the time slot Sun for the uplink signal is an empty slot, it is decided to use this empty time slot, and the necessary timing is determined from the response signal from the wireless base station 30 by sending out a control signal for calling. It is also possible to send out burst-like control signals.

If a call is made from a 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 it finds an empty time slot in which 100 mobile radios can make a call, instead of using the entire time slot, some mobile radios use only the first half, and some mobile radios only use the second half. This is a method that allows you to use 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). Find the time slot that is currently in use. The time slot given to the mobile radio device 100 may be Sun, but a search is performed just to be sure. It is mobile radio 1
In addition to 00, it is also used to respond to simultaneous calls from other mobile radios, and to provide service types and time slots appropriate for the service types.

As a result, if time slot SD1 is vacant, for example, time slot SD1 of radio channel CH1 is used for mobile radio device 100, and time slot uplink (mobile radio device 100 → 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 designated time slot SD1, and at the same time shifts to a time slot 5U1 for the uplink radio channel corresponding to the downlink time slot SD1 (FIG. 2A). (b)). At this time, the control unit 140 of the mobile radio device 100 operates the transmission/reception intermittent controller 123 and starts operating the switches 122-1 and 122-2 (8204>. At the same time, a slot switching completion report is sent to the uplink time slot SU1. 5205> and waits for a dial tone (
S206>.

Time slot SL of the radio carrier of this upstream radio signal
The +1 state is schematically shown in FIG. 2B (C). In addition to the time slot SU1, the radio base station 30 also receives SU as an uplink signal from another mobile radio device 100.
3 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 gateway exchange 20 820B>, upon receiving this, the gateway exchange 20 detects the ID of the mobile radio 100 and selects the necessary switch from among the switch group included in the gateway exchange 20. Turn on (320
9>, send dial tone (3210, 4th B)
figure).

This dial tone is transferred by the wireless base station 30 (S211>, and the mobile wireless device 100 confirms that a communication path has been set (3212>).

When transitioning to this state, telephone unit 1 of mobile radio m1oo
You can hear a dial tone from the 01 receiver, so start sending out the dial signal. This dial signal is speed-converted by the speed conversion circuit 131 and sent out from the wireless transmission circuit 132 including the transmission section 134 and the transmission mixer 133 using the uplink time slot SU1 (3213>. Thus, the transmitted dial signal is It is received by the radio receiving circuit 35 of the base station 30.

This radio base station 30 already responds to the calling signal from the mobile radio 100, gives the time slot to be used, and operates the signal selection circuit group 39 and signal allocation circuit group 52 of the radio base station 30. Let's go up the time.
Receive slot SU1 and receive downlink time slot SD
1 signal is being transmitted. Therefore, the dial signal transmitted from the mobile radio Ifi 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. Call signal 22 via communication @ processing unit 31
-1 to the gateway exchange 20 (3214>, and a call path to the telephone network 10 is set (3215>).

On the other hand, an input signal from the barrier switch 20 (initial control signal,
When a call is started, the call signal) undergoes speed conversion at the signal speed conversion circuit group 51 at the wireless base 830, and is given a time slot SD1 by the signal allocation circuit 52-1 of the signal allocation circuit group 52. There is. Then, the time slot SD of the wireless channel downstream from the wireless transmission circuit 32
1 to the mobile radio 111100.

The mobile radio device 100 is waiting for reception in the time slot SDI of the radio channel CH1 and is received by the radio reception circuit 135, the output of which is sent to the speed recovery circuit 138.
is input. In this circuit, the original signal of the transmission is restored and input to the handset of the telephone section 101. Thus,
A call is started between the mobile radio device 100 and a regular telephone within the regular telephone network 10 (S216).

The call is terminated by turning on the receiver of the telephone unit 101 of the mobile radio device 100 (S217>), and transmitting the call end signal and the on-hook signal from the control unit 140 to the radio transmission circuit 132 via the speed conversion circuit 131. Wireless base station 30
(3218>, the control unit 140 stops the operation of the transmission/reception intermittent controller 123, and fixes the switches 122-1 and 122-2 to the output terminals of the synthesizers 121-1 and 121-2, respectively. .

On the other hand, the control unit 40 of the radio base station 30
When the call termination signal from 00 is received, the call termination signal is transferred to the barrier exchange 1120 (3219), and the switch group (not shown) is turned off to terminate the call (S220).
. At the same time, the signal selection circuit group 39 and signal allocation circuit group 52 in the radio base station 30 are opened.

In the above explanation, control signals between the radio base station 30 and the mobile radio device 1OO are controlled by the signal speed conversion circuit group 51. Although it was explained that the signal speed restoration circuit group 38 etc. are not passed through,
This is for convenience of explanation, and communication can also be carried out without any problem even through the signal speed conversion circuit group 51, signal speed restoration circuit group 38, control signal speed conversion circuit 4B, or signal processing section 31 in the same way as with audio signals. .

(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 state of waiting to receive the downlink control signal of the radio channel CH1, which is desired to be followed by the procedure determined by the system.

Assume that an incoming call signal to the mobile radio device 100 arrives at the radio base station 30 from the general telephone network 10 via the barrier switch 20. These control signals are transmitted as communication signals 22 to the control section 40 via the signal rate conversion circuit group 51 and the signal allocation circuit group 52, similarly to the voice signals. Then, the control unit 40 selects an empty slot among the downlink time slots of the radio channel CH1 addressed to the mobile radio device 100, for example, S
ID signal of mobile radio 100 using Dl, native call signal display signal, time slot use signal (mobile radio
For transmission from 0, for example, SU1 corresponding to SD1 is used. Mobile radio device 10 that received this signal
0 is transmitted from the receiving section 137 of the radio receiving circuit 135 to the control section 140. The control unit 140 confirms that this signal is an incoming call signal to its own mobile radio 100, so it makes a ring tone sound from the telephone unit 101 and at the same time calls the designated time slot SD1. The transmission/reception intermittent controller 123 is operated to stand by at SU1, and the switch 122-1.1222 is started to be turned 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 performed in good condition using this system and does not adversely affect other wireless channels in the system, so it will be omitted here. A theoretical explanation of the TCM signal having multiple load gain applied to the present invention will be given, and then its application will be described.

(3) Regarding the multiplexed gain of the TCM signal transmitted from the radio base station 30, to associate the multiplexed gain of the TCM (time division time compression multiplexed) signal with the multiplexed gain of the FDM (frequency division multiplexed) signal. , First, "Each channel CH1 of DM
, CH2, . . . each audio signal flowing through CHn is expressed in the form of a function. The FD~1 signal is a signal obtained by converting the frequency of an audio signal and arranging it in a line on the frequency axis as shown in Figure 5. This multiplexed signal is used for coaxial transmission system or microwave analog communication system. It is widely used, and multiple load gain has also been incorporated into practical systems to great effect. Hey, the spectrum in Figure 5 has 12 channels.
(CH1-12), but in general, 12
In addition to 60,120,480.960.1200.2
A wide variety of 700 pieces are used.

In the following explanation, the FDIVI signal or DCM
It is assumed that the input audio signal levels to the signals are the same.

Now, the channels CH1, CH2, CH3 in Fig. 5...
..., the audio signal flowing to CHn (in the case of wired, the frequency band to be transmitted is 0.3 to 3.4 KHz, but in the case of mobile radio signals, it is 0.3 to 3.0 KHz, so this value is limited〉 as f (t), f 2 (℃),...
..., fn(1). The frequency components of these signals are channel CH1 or 0.3~3°0KH2,
CH2 or 4.3~7.0KH2,...

CHn is 4X (n-1> +0.3~4X (n-1
>KH2, and there is no overlap with each other.

However, when viewed from the signal waveform, the imaging width distribution of these n audio signals is simply shifted to a higher frequency on the frequency axis, and the signal 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 of the FDM signal is the signal when n voices are arranged on the frequency axis as shown in Figure 5, and the signal when nfl without frequency conversion.
This is exactly the same as simply mixing the voices of ii. Prove this using a formula. Channel CH1, CH2,...
...The mixed signal of CHn is expressed by the following equation.

F (t) = f1 (1) + f2 (t) + - + fo < t) (
1> Specifically, f・(1) is expressed as follows.

(2) (4i-1)kHz (3) However, 1≧2 Furthermore, the mixed signal when frequency conversion is not performed is given by the following equation.

G(t) =C] 1(t) 1002 (t) 10・=”
gn (r) (4) Here, glft)=f1(t) kH2 (5) (6) However, 1≧2 Next, the power possessed by the signals of equations (1) and (4) is determined.

First, the power of F (t) is (7) On the other hand, the power of G (t) is (8) However, we used the fact that no power is formed between different signals. That is, f, dt=0 ■ f Cli Clj dt=0 fa ・s + n (w p i βpi) op+ xa −5jn (ωp3+β, j) dt=01) J (9) However, i≠j Now, From equations (2> and (3), (10) (11), however, (≧2) From equations (10) and (11), F(t)2miG(t)2 (12> is obtained.

In other words, it has become clear from the above explanation that the power of the signal is not related to frequency conversion.

Next, consider applying the sampling theorem to C11(t). Since the highest frequency fh of gi([) is 3KHz, the time interval 1/(2f1.,), that is, 1
It is well known that if sampling is performed every /6000 seconds, the original signal can be reproduced later even if only the sample value (voltage value) is transmitted.

Therefore, f・(1) is expressed as time interval +j/6000+ (1/6000)(i -1),' as shown in Figure 6A (a), respectively.
Sample every 6000j seconds (13). In the same figure, ta=1/6
000 seconds.

t b= (1/6000) x (1/6000)
seconds.

to= (1/6000) X (5999/6000
) seconds.

td=1/6000+ (1/6000) x (30
00/6000) seconds.

te=1/6000 seconds.

It is. Hereinafter, in order to explain specifically, the multiplex number n is set to 6.
000. The length of one frame is 1/6000 seconds.

Now, on the horizontal axis of Figure 6A (a), place 600 small bags (diameter 1/6000 x 1/6000 seconds) as shown in Figure 6B (c).
0 pieces, and one large bag with a diameter of 1/6000 second, are lined up as shown in the figure. Then, we compare the above sampled values to matchsticks and consider how these matchsticks fit into the bag.

Function CJ・(t) (i=1.2.-, n)L, 11
Since I own 6,000 Matsushi sticks for each second and second, and they are spaced at equal intervals in time, I have a bag (1), (2),...
2(N) is filled with one line each, and this operation is
It will be repeated every frame. That is, bag (1) has Q 1 (1,/6000) bag (2) has Q 2 (1/6000+1./6000x 1/
'6000) Bag (3) l, :ha, ni 13 (1/60
00+1/'6000x 2/6000) bag (6000
), Q (1/6000 ÷ 1/6000000 X5999 /6000) will be entered respectively.

Also, in the large bag (Σ6000), the mixed signal G(t
) is included. In this case, the sampling time is 1/6000 to 1/
6000x3000/6000, that is, the midpoint of one frame. Then, the value of the next matchstick will be stored in the large bag (Σ6000).

The large bag (Σ6000) has G (1/6000 1/6000X3000 /60
00) Below, 1 shift of match sticks from bags (1) to (6000), 11 (1/6000) Ω2 (1/6000 to 1/6000X 1/600
0).

Q 6000 (1/6000+1/6000x 59
The sum of 99 /6000) and the value of Matsushi's salary in the large bag (Σ6000) (1/6000 + 1/6000X 3000 /
6000) are compared as shown in equations (14) and (15) below.

000 Σgi=(71(1/6000) + g 2 (1/6000+ 1/6000x 1
/6000)! Q 3 (1/6000-4-1/6
000X 2/6000)+・・・-・+C1(1/6000+1/6000x(i-1)/6
000)-g6ooo (1/'6000-t1/60
00x 5999 /6000) (14) G (1, /'6000 work 1/6000x3000/6
000 ) = J (1,'6000-1〆"600
0X3000/6000 )+Q2 (1/60
00-=1'6000X 30. -Do/6000
)-8-Q n (1/6000: L/600
0X 3000/6000 ) (15> As a result of comparing equations (14) and (15), if it is 000 (16), then the average value or If it converges, the multiple load gain in the FDM signal is T CM (also in No. 8, it has been proven that the same value exists. This is because 6° placed on the horizontal axis 00 bags, the bag represents 1 time slot, the total bag is 1 frame, and the matchstick inside the bag has each signal q, Q2
．． ......, Ω□ is time division time compression multiplexing (Ding CM>
The TCM signal can be considered as the signal shown in Figure 6B (c).
This is because the signals are well mixed, like a large bag (Σ6000), and can be considered as one FDM signal. Therefore, in this example, even though it is a TCM signal, there is no particular need for time compression, and the degree of compression is 1.

In addition, in Figure 6B (C), the position of the bag on the horizontal axis (time axis) for bag (1) is 1/6000≦↑<1/6000+1/6000
x 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 bag (6000) is 1/6000-L1/6000x 5999/6000
≦↑＜ 1/6000+1/6000x6000/60
00 is shown. On the other hand, the audio signal q・(i) , Q2 (j) ,... gi
(t) ,...q6000(1) The sampling time is 1/600, respectively.
0. 1/6000+1/6000X 1/6000.
......, 1. '6000+ 1/600
0X (1-1)/6000.・・・・・・、
Since the setting is 1/6000+1/6000x5999/6000, each matchstick will touch the side of the bag and then enter the bag. This is done for convenience; if you want to put a matchstick in the center of the bag, for example, change the time t of q・(1) to t = 1/6000+ 1/6000X (i+ 0.
5) You can select something like /6000. 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)).

Δ・ = C1i (1/6000+1/6000x (
i-1)/6000)-gH(1/6000+1/60
00x3000/6000) (17) What the above formula means is that when sampling the audio of the i-th channel (CHr >), the time is (1/6000+(i-1)/6000") seconds, and 60002) represents the difference in the signal magnitude between seconds.This difference is like the error value of random noise, and represents the difference in signal magnitude between channels (CHi) i = 1, 2.-...-, n. , a positive value, a negative value, or
Occasionally, it may be possible to get O, but in boat fishing there will be variations left and right around O, and the variations will be normally distributed.

The above was about a certain time 1=10.

The next sampling takes place 1/6000 seconds later.

The next one is delayed by an additional 1/6000 seconds. For example, this sampling is performed 6000 times per second, so the average value of equation (17) for one second will be (18), that is, it will approach O(. It becomes even clearer if we multiply the equation (18) by taking the average over 10 seconds or 30 seconds.

As described above, FDM and T with the same multiplicity (6000)
Although it has been clarified that the average power level of each CM signal is the same, the amplitude distribution of the above two types of multiplexed signals will now be explained.

In FDM, if the standard deviation of the amplitude distribution of a single audio signal is σ, then in n multiplexing, it is n1/2σ, whereas in TCM, the signals are not mixed, so the standard deviation is the same as the original signal. Therefore, when transmitting a TCM signal, the peak value of the signal is smaller than that of an FDM signal, but it is still a TCM signal.
The highest frequency of the signal is 3KHz x n times higher, so if you want to increase the modulation deviation of FM modulation and cause interference to adjacent channels, for example, change σ to σ in advance.
A circuit for compressing to 1□′2, that is, 1/2 compression (
It is necessary to insert a compressor 171 having the function of increasing the decibel value to 172 as shown in FIGS. 1D and 1F and then adding it to the transmitter. On the receiving side, there is a circuit that performs the expansion operation in reverse, that is, an expander 17.
2 as shown in FIGS. 1D and 1E, it is necessary to reproduce the original sound.

The above shows that the multiple load gain obtained by FDM is approximately CM
This is nothing more than showing that it is possible to change the amplitude distribution of the signal 1q so as not to interfere with adjacent channels. However, the value of the multiple load gain 1q quoted from the above-mentioned document 3 is that the frequency band of the audio signal is 0.3 to 3.4 KH7, whereas the above value is defined by Japan's Radio Law Enforcement Regulations. It was assumed that the same value could be obtained in the audio transmission band of 0.3 to 3.0 KHz.

This assumption can be accepted with virtually no error.

In the case of the TC~1 signal, since the signal is time-compressed, its frequency components become more compressed, but as mentioned above, only the frequency components are changed, and the signal waveform itself is Since it is only subjected to a similarity transformation extending on the frequency axis, there is no change in the amount of multiple load gain, but this will be proven below using a mathematical formula. ' The signal C~1 is expressed as follows using equations (5) and (6).

3,0K)lz (19) However, ffT<t<T/n-! l! Th(t)=○ However, 1/n〈T〈↑〈T+Tf=1.2,
3.・・・・・・ Below, the time-compressed total TO~1 signal, which was once one frame long, is (20
) The reason why the value is multiplied by n1'' on the right side of equation (20) is that the TCM signal is only transmitted for a time of 1 y''n within the frame time. Mouthiness is expressed in terms of voltage (in terms of electric power, it is n). Now, if we calculate the power in equation (20) for the time T of one frame, we get the following equations (7) and (8): (21) Therefore, if we take the time to be an integral multiple of the frame, the signal H
(The power possessed by H and G(1) is H(t)?
Three G(t)”
(It turns out that 22>.

First, we will explain the multiple load gain when the audio n-channel multiplex TC is shorter than the frame length of one signal (1/12fh).

In this case, as above, FDM of audio n-channel multiplexing
It can be easily proven that it has a value equivalent to the multiple load gain applied to the signal.

For example, assume that the frame length is 1/8000 seconds. Then, the sampling frequency was changed from 6000H2 to 8000H2, and each audio signal g1m, Cl2(
t), . V, that is, Figure 6A (a), (b), Figure 6B (C)
The above explanation can be applied simply by changing the horizontal axis from 1/6000 to 1/8000.

Furthermore, the multiple load gain and growl will be explained when the frame length becomes longer than 1/(2fh). In conclusion, the multiple load gain generally decreases, and its specific value is determined below. As a concrete value, the number of multiplexes is 6000, which is the same as the previous example, and the frame length is 1/300.
An example of a case where the time becomes 0 seconds will be explained. The size of the bags arranged on the time axis increases by 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 all the diameters of n bags, we get 1/300
It becomes 0 seconds. Also, the large bag (Σ6000) is about 1/3 the diameter.
000 seconds.

Now, as before, the sampling frequency is 17'60.
Consider putting the Matsuchi stick sampled at 00 seconds into a container bag. In this case, the aforementioned frame length is reduced to 1/600
If you set it in exactly the same way as when you set it to 0 seconds, the following inconvenience will occur. That is, as shown in Figure 6B(d), during 1/6000 seconds, the bag becomes (
There are only 1) to (3000), while Matsushi pay is 6
Since there are 000 bottles, each bag can contain two bottles. In other words, as shown in Figure 6B (d>), bag (1) has audio signals Ω1(t) and Q2 (t)9, bag (2) has audio signals C12(T) and Cl3(t'), and bag (3) has c+
3m and Q4 (i) ,..., hereafter, bag (30001
The four matchsticks (signal values) representing g3000[) and 03001(t) are inserted. Note that Ω1 (
t) is the previous sampling time 1 (6000)
It is placed in On the other hand, bags (3001) to (6000
), the number of Matsushi sticks sampled within this time is 1.
I couldn't even put a book in, and it was time for the next sampling, 1.
Two matchsticks sampled within /6000 seconds will be placed in each case.

On the other hand, the large bag (Σ6000) has two matchsticks in one frame, so two matchsticks, that is, G (1/6000-!-1/6000X3000./
6000) and G (1/6000-!-1/6000
X9000/6000) will be included. However, G (1/6000+1/6000x 9
000/6000) is the matchstick (signal) sampled within the next 1/6000 second, as described above.

What kind of goods does the above refer to?

That is, the fact that two matchsticks are put in bags (1) to (3000) means that each time of the signal C to C1 is
Audio CJ (t), “i+1ft)” in slot 2
This means that they should be mixed channel by channel. To do this technically, two-channel (CH) FDM is required.
2 (CH) x 3 for a total of 3000 slots.
This means that a CM signal of 000=6000 (CH) should be generated. And these and a large bag (Σ6000
) will be compared in size with the previous match stick in the same way as described above.

Although this is also a method, it is not a TCM signal in the original sense. Therefore, in order to include only one audio signal in each time slot, the following must be done. 6000 audio signals are divided into two groups, one group can be put into bags (1) to (3000), and the other groups can be put into bags j3001) to (6000).

This operation will be explained using FIG. 6D (f).

There are bags (1) to (6000) and large bags (Σ600).
0) or two sets of each are prepared. Now, the bags (1) to [6000] with the writing 7 on the top contain 11g of matsushi sticks.
3゜q5. ..., ``5999'' or one bottle each.And a large bag (Σ6000)
is G1 (1, I'6000 11.'6000X3
000/'6000) There is one matchstick shown. On the other hand, the bags written at the bottom (1) ~ (6000
) is Matsuchibō C]2. C]4. Q6゜..., q60
00 or one each, and a large bag (I6000
) L:HaG2 (1/6000+1/6000x9
000/6000) is displayed.

By doing as shown in this figure, two match sticks will not be mixed in the same bag, and with the same proof as in the case of 1 frame length or 1/6000 seconds, FDM
It can be seen that the signal multiplexing interest rate is 1% (in this case, 3000 multiplexes), which is the same as that of the CM 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.

G1 (t) −C]1 (t) 7g3 (tJ+・
...-7-g5999(1)(4') G2(t)-g2(1)shiq4(1)-...+g6o
oO(t)(4'') Then, each of the above equations can be proven by performing the above-mentioned proof. However, the sampling timing is assumed to be performed under exactly the same conditions as above.

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 in the case of TCM signals, signal compression is necessary.

In Fig. 6D (f>), the upper bag group (1) to (6000) and the lower bag group (1) to (6000) contain only one matchstick, but the TCM signal However, there is still some dissatisfaction.The reason is that the upper group of bags and the lower group of bags are progressing at the same time, and therefore, within one time slot of the D C and Vl signals. is still ql and G2.

g3 and G4, ``'', 5999 and g6000 are supposed to coexist. If you want to remove this or compress the signal, use the following method. In other words! J
, g3 ,... Regarding 1g No. 5999, there are two bags (1) and (3001), (2) and (3002),
... [3000) and (6000) are stored in the previous bags (1), (2), ..., (3000>, respectively, and G2, g4, ...2g6ooo are stored in the same way. Later bag (3001+, (3002)
,..., (6000) [this is accommodated.

To do this, for example, the following composite signal may be created for gl. That is, it is sufficient to create the sum of signals obtained 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. It is desired. However, in this case, the value (voltage value) of the CM signal at the instantaneous value of the sampling time is
It is impossible to faithfully reproduce the original signal, and it becomes necessary to transmit a signal with a fixed time width (time slot length).

If we look at the above operation from the perspective of multiple 9-carrying gain, it will be as follows.

If the frame length or sampling time interval is longer than 1/6000 seconds, 600
The multiple load gain in FDM for 0CH multiplexing can be obtained.If the frame length is 1'3000 seconds, the multiple load gain is equivalent to the multiple load gain for 3000CH FDM signals.

Furthermore, the frame length time is increased to 1 second, and the multiplexing gain is calculated when the number of multiplexing n = 6000.

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 g1(t),...
``6000 (mouth), respectively, and consider where the sampled matchsticks can be placed.If the sampling timing is the same as in the case of the frame length or 1/6000 second described above, then the bags (1), (2), ( 3).

. . . (6000), one matchstick indicating each audio signal will be inserted into each one.

On the other hand, in the large bag (Σ6000>), there are 6000 matchsticks (having values sampled in each sampling period) indicating G(1).This means that in each time slot of the CM signal, This means that 6000 channels of audio are mixed and input.The technology that makes this possible is FDM, but this is not suitable for the purpose of this application,
(It cannot be applied to Mg. Therefore, in order to input only one audio signal in each time slot, the 6000 audio signals must be divided into 6000 groups, one group at a time (in this case, one channel). one by one).

This shows that the multiple load gain in this case is O, which is not obtained at all. Further, the signal compression degree of the CM signal in this case is 6000.

As is clear from the above explanation, we have shown that the multiple load gain decreases when the length of one frame becomes longer than the sampling period required to faithfully reproduce the audio signal. If expressed as , the frame length t8 becomes t
e > 1/(2fh), and if the number of multiplexes is n, the multiplexing gain is n' = nX 1/(2fh
te) (23) This value is equal to the multiple load gain of a frequency division multiplexed signal having a multiplex number determined by the following value.

Frame length t. According to the above-mentioned documents 1 and 2, the practical range of the multiplex number n is approximately 3000≧ n≧2, where the frame length t: o, i seconds ≧ 18 ≧ 0.0005 seconds, and the above-mentioned It will be clear from the above example that as long as the value is within the range, equation (23) always holds true.

In addition, in order to input one matchstick into the multi-band as a TCM signal, in other words, to input one original audio signal, the time length of one frame must be longer than ↑ 8 or 1/2 fh. It has also become clear that in the case where the signal must be compressed, the compression ratio ψ is given by ψ=te/(1/2fh).

Roughly speaking, the variance of the amplitude distribution of the TC~1 signal is the same as the variance σ2 of the telephone signal, as explained below.

Here, the signal f(e)(
t'=1.2.3. Assume that the mean value and variance of (...) are as follows.

Average value E[f(ff) ko O 2° Dispersion E[f(ff)'' = σ2 As is well known, the average value and variance of N multiplexed FD~1 signals are as follows: E [Σf −(n) 2E =la2 1=1 Therefore, they are not arranged like the kmFDM signal on the frequency axis, and the above-mentioned function Q・<r=1.2゜...
. . , n) is similarly obtained. The average value is E ``f(n) co-ffim(1/' rN
)r→(3) −1 Dispersion Next, the average value and dispersion of the CM signal used in the present invention are determined. In this case, the function f-(n + k
) is renamed according to the following relationship.

f i (n+k) = f (kN+n+
1-1) Now, find the average value.

[-1 E [f(n)] = pim1. /1−Σ f (
n) [→oo n=0 For simplicity, L=rN =Σ 1/beim (1/r) j−>(su−1 =E [f・ (shi)] seek.

L-)oO n=O For simplicity, let 1--r\, then 2] EIf(n), -ffim (to 1/r) r-+■ = Σ 1/N eim(1/r ) j-+(X) From the above, it has been proven that the dispersion is the same as that of one channel of audio signal.

(4> Regarding the multiple load gain of the CM signal received at the radio base station 30 The radio base station 30 receives TC1-1 signals transmitted from a large number of mobile radios 100. Let us consider the multiple load gain 1q possessed by the wave. To conclude, as will be described later, assuming that the mobile radio 100 can obtain exactly the same multiple load gain as when transmitting from the radio base station 30, the modulation factor You can see if it is OK to send by deepening the message.

However, the wireless base station 30 has a compressor (¥f-) 71
When used as shown in FIGS. 1F and 1G, a compressor 171 is also installed on the mobile radio device 100 side as shown in FIGS. 1D and 1E, and a wireless transmitter circuit 132
Is it necessary to insert it before the ?

As a specific example, 1 frame length is set to 1 sampling time interval.
/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 6,000 time slots of one frame.

The mobile radio devices 100 are arranged at equal intervals 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 omnidirectional. And the magnitude of the transmission power from each mobile radio device 100 is all the same, and each mobile radio device 100
The carrier waves used for transmission are assumed to be phase synchronized with each other. Furthermore, the radio wave propagation characteristics between the mobile radio device 100 and the radio base station 30 are as follows.
and the wireless base station 30 are identical. Under the above assumptions, the transmitted signals of 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 100 is transmitting only a single voice channel in its given time slot, the number of multiplexes 600 can be obtained from the multiplex load gain.
This shows that it is possible to transmit using a modulation depth that incorporates a multiload gain of zero.

Although ideal conditions have been set above, consider the actual system operating conditions. In this case, the positions of the mobile radio devices 100 are randomly scattered, and the radio wave propagation state changes in various ways, so the received power of the radio base station 30 varies for each time slot. Furthermore, the carrier waves from each mobile radio device 100 are not necessarily phase-synchronized.

Therefore, if the depth of modulation is large in time throwers 1 to 1 where the reception level is high, it is expected that the adjacent time slots will be adversely affected by the influence of multiplex radio wave propagation. However, this can be alleviated by taking other measures such as increasing cart 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 located in different locations; on the contrary, there is a possibility that this can be used to repeatedly reduce the number of zones. be. This is because multiple load gain is used as FM (PM) modulation, and as a result of deep modulation, wideband gain is obtained, and the resistance to co-channel interference is increased.

Taking all of the above into consideration, the system design assumes that substantially the same multiple load gain can be obtained when the mobile radio 100 transmits and when the radio base station 30 receives, as when transmitting from the radio base station 30. It became clear what could be done.

(5) Specific utilization method of multiple load gain The multiple load gain is determined when one frame length is 1 m5ec and the number of multiplexed CM (time division time compression multiplexing) is 500, and an example of its utilization will be explained.

First, 5CPC (Single Channel Pe
r Carrier.

The signal of one telephone channel is modulated on one carrier wave. Find the analog FM signal S to "Ii sound\ratio and compare it with TCM.The level of the input signal to the receiver (
The voltage value) is C, the FM modulation index is mf, and the noise level per unit frequency (voltage value) is n.

Assuming that the bandwidth of the low frequency amplifier in the discriminator output stage is Fa, and the highest modulation frequency f of the modulated wave is equal to Fa, the signal-to-noise ratio is given by the following equation.

1/2　　　　　　-1/2 S/N=3 mrC (2F8) (24) In addition, this formula was quoted from the following literature.

Edited by Sugawara “FM Radio Engineering °゛” Published by Nikkan Kogyo Shimbun, 1952
Page 401 (13, 25) Formula Next, the TCM signal with multiplexing number Q is FM under the following conditions.
The value of S/ in this case is given by the following formula.

(25) However, F, o=QFa "fo-Qmf (25') n O= Q n In other words, in the TCM signal, the frequency of the original signal is multiplied by Q, so the bandwidth of the low frequency amplifier is multiplied by Q. The modulation depth (modulation index) has also increased by a factor of Q, and the noise level has also increased by a factor of Q, so it is appropriate to use equation (25').

Next, the TCM reception signal level 3172mfo to take the same value for S/ on the left side of equations (23) and (24)
co(2Fao)-172(26) From formula (26), we obtain C,=Q"C(27>.In other words, the reception level is 01" in voltage.
This means that 0 times the power level is required. Therefore, it is necessary to increase the transmission power by Q times from 5CPC.

Next, find the multiple load gain of the TCM signal in the above example.

As explained in section (3), in this case FDM
The equivalent multiplex number is n' = 500xl/6000 (sec >-(1
/1000 (sec)) = 500xl/6 =83
It can be seen from FIG. 11 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.

When the graph shown in FIG. 10 is created and estimated based on FIG. 11, a multiple load gain of 30d13 is obtained. Therefore, this multiple load gain is used to deepen the modulation depth (deviation) and reduce the transmission power. 5cpc without TCM
, In other words, if the transmission output of a 1-channel analog FM signal is iomw at the cordless telephone level, the required transmission power in this case can be found by multiplying by 500 using equation (27) and then subtracting the multiple load gain, 10 log (10mWx 500) -30d Bm
7d Bm(28) That is, we get 5.0mW. This means that using TCM requires less power.

First, we will explain the physical meaning of the multiload gain in the C-1 signal, and discuss points to keep in mind when using this as a system.

As mentioned above, if the frame period of the TC and M signals is long and becomes more than 1 second (assuming the number of multiplexes is 6000), the multiplex constraint profit of 1* cannot be obtained. The F, ``V1 modulation index has a fixed value determined by the system.For example, the original signal (03~3.OK+
The modulation index of -12> is 1.75 KH2 (in the case of a tone signal of 1 KH2 and standard modulation deviation), which is 500
The signal band for multiplexed TCM is 150-1500K
Hz, the standard modulation deviation is 875KH2 (50
(When the 0KH2 tone signal is standard modulated). However, if the frame length is set to 1m5ec, a multiple load gain of 30 dB can be obtained as shown in the above j.
Although the tsubo was used to increase the depth of modulation, we will explain how the modulated wave actually behaves.

Consider an all-channel implementation, ie, a telephone signal is flowing in every time slot. In this case, the influence of the multiple load gain of 30 dB on the modulation deviation increase is, considering from Figure 12, the multiple load gain or the relative power of the sine wave whose peak value is equal to the value minus about -L-8 dB. CM with arbitrary frame length.
It can be seen that this is equal to the modulation shift of the signal when only one time slot is used in the signal.

Next, consider the case where the load is gradually reduced from all time slot implementation. That is, the question is what will happen to the modulation shift of the signal when some percentage of the time slots are used for actual audio signal transmission and the rest are not used as empty time slots.

In this case, since the number of installed channels is reduced, the multiple load gain is naturally reduced as well. For example, 1/2 25
0 implementation, the multiple load gain is n' = 250X1/6000+ (1/1000>
= 250X1. /6 = 42 (CH) once,
It can be seen from FIG. 10 that the multiple load gain is 245 dB. However, since the number is 9 or 172, the power level of the modulated signal is lowered by 3 dB. Therefore, the geometric multiple load gain in this case is 27.5 dB, and T
Although the effective modulation depth of CM-FM is slightly larger, this is considered to have no effect on system operation.

Find the effective modulation shift when the number of implementations is roughly reduced and only one time slot is used. The multiple load gain 1q for one channel is OdB, but the signal load is reduced by 27d3 by 11500 when fully mounted. Therefore, the apparent multiple load profit 1q is 27
dB, and it may be assumed that even if the modulator is operated at 30 dB, there will be no effect on system operation. Also,
In addition to the above-mentioned compressor, the input stage of the modulation circuit of the actual radio equipment includes an IDC1n5tantaneous D
An eviat+on control (instantaneous modulation shift amount suppression) circuit is provided, and a function is provided to limit 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 telephone signals of C-1.

As explained above, the multiplex 9 of the signals C1-1
It has become clear that by using the signal gain to increase the modulation deviation of the ~1 signal, it is possible to significantly reduce the transmission output.Technically speaking, this has a very large effect on power saving. It means that.

In other words, a 10 mW continuous transmission radio at 5 cpc is operated at a time rate of 11,500, or 0.2%, and its output is 1. /? It is obvious that the power saving effect is great since only 5 mW is required.

Next, a system comparison is shown in FIG. 7 when the frequency bands of cordless telephones currently used in Japan are used in the CM system, and the present invention will be explained in detail.

In FIG. 7, various types of CM systems are listed for comparison of various characteristics of cordless telephones and FDM systems. Here, the abbreviations of each specification will be explained.

fo...Carrier frequency NCH...Number of required wireless channels n...Number of possible calls (calls) N...Maximum number of calls as a system max fw...Frequency band of modulation signal ■m... Maximum voltage at modulator input stage (1/'60
Average for 00 seconds〉 q'...Multiple load gain (estimated from Fig. 10)〉Vqm・
... Maximum voltage at the modulator input means when multiload gain is added (1/6000 second average) Md ... Modulation depth (standard modulation) Pl ... Transmission power A3 ... Service area ( distance from wireless base station)
t8... 1 frame length "TS... Number of time slots in 1 frame n'.
...The number of multiplexed FDM conversions (excluded from equation (23)) In Fig. 7, the TCM system systems 1A, 1B, 2.
3, the maximum number of calls N as a system is calculated from Reference 1 by using the frequency band 12.5 kHz x 89 CH = 112.5 kHz given to the one-way station max of the cordless telephone with the TC~1-FM signal of one carrier wave. , 148 multiplex, i.e., cordless telephone replacement 14
It can be seen that up to 8 channels (CH) can be used.

In the same figure, the frequency band fw of the modulation signal of the D~1 system was calculated as 444kHz for system comparison.Also, the multiple load gain q' of TCM system 1B is OdB when calculated from equation (23), but the system For comparison with 2, the figures in parentheses are included.

Multiplexing number of TC~1 system 2 in Figure 7 (number of time slots NTs in one frame) 148, frame length t
Find the multiple load gain of the TCM signal of 8=1 m5ec. As shown in equation (23), in this case, the FDM equivalent number of multiplexes is 24.6 CH1, so from Fig. 11, the multiple load gain is an intermediate value between 28.6 for 600H multiplex and 32.6 dB for 1200H multiplex. I know that.

Estimating from FIG. 10, we get 21 dBjr. Therefore, this is used to deepen the modulation depth (deviation) Md and reduce the transmission power P. 3 CPs not using TCM
C (Single Channel Per Carr
ier), that is, if the transmission output of a 1-channel analog FM signal is 10mW, which is the level of a cordless phone,
The required transmission power in this case can be found by multiplying by 148 using equation (27) and then subtracting the multiple load gain.

That is, 10 log(10mWx 148) -21dB=1
Obtain 1.7d13m. For those who have converted to TCM, the time rate is 1/1.
The transmission time will be 48, and the transmission power itself will be approximately the same. However, depending on the operating conditions of the system, it may be necessary to include a compressor 71, 171 with a compression rate of about 172 in the transmitter input stage to suppress the peak output.

The TCM system shown in Fig. 7 cannot obtain a large multiple load gain q' like the FDM system, so it is inferior to the FDM system in terms of gradual reduction of the transmission power P.
Since the mobile radio device 100 of the CM system performs intermittent transmission using time slots, its average power is FD
It will be on the same level as the M system. In addition, taking into account that the hardware configuration of the mobile radio m1oo of the TCM system is very simple and inexpensive compared to the mobile radio of the FDM system, the DCM system is superior to the current one. It has become clear that this system has higher frequency utilization efficiency, can save power, and is more economical than both cordless telephone and FDM systems.

Note that in the TCM system, the shorter one frame length is, the greater the multiple load gain can be obtained, but it cannot be made too short. That is 1 frame length ↑ for an audio signal. If the length is too short, the number of samples cannot be increased, making it difficult to faithfully reproduce the original sound. Practically, one frame length 1o is about 1 to 5 m5ec.

(6) Physical meaning and considerations of multiplexed 9-weight gain in CM signal The CM system shown in FIG. 7 will be compared again. system 1
In A, each time slot S as shown in FIG. 8(a)
The signal of 1 second of audio is compressed to 1/148 second,
After being FMed, it is sent out into space, whereas in Fig. 8 (b
) system 2, the audio signal of 1/1000 seconds is further multiplied by 1/148, subjected to FM, and then sent out into space. Hereinafter, in System 1A, the modulation depth Md must be maintained at 1.751" ad rms, similar to a cordless phone, whereas in System 2, this must be maintained at 21 dB.
Let's consider the physical reasons why it is okay to deepen the depth.

First, in system 1A, the transmission power P1 is lowered by 21 dB than the value shown in Fig. 7, and the modulation deviation (modulation depth) Md is increased by 11.2 times (21 dB) in systems 1B and 2. Which comparison to consider. However, if the IDC (Instantaneous Modulation Deviation Amount Suppression) circuit or compressor 1
71. It is assumed that expander 172 is not used in both system 1B and system 2. If you do not take into account the influence of radio wave propagation characteristics such as multiplex propagation on communication, but if you pay attention to other design specifications, it will be possible to It is necessary to be able to receive data with the same reception quality for both systems.

Next, we examine the influence on adjacent wireless channels. system 1
In B, there are many cases where an audio signal, such as "necessary", is transmitted at a high level for 1 second, and therefore TC~
Even if the modulation deviation is 1/148 seconds after 1, for example, 1
．． Sending the modulated wave into space by making it 11.2 times larger than 75 rad rms reduces the transmission power to 11.7 m.
It will be demonstrated below that even if it is reduced to Δ, it will cause significant interference to the adjacent channel or to the communication of the host.

Quantitatively determine whether the system is larger than system 2. For this purpose, it is sufficient to compare the signal powers of systems 1B and 2 that exist within a certain period of time. However, setting this time too long would be meaningless, so here it is set to 17148 seconds. Then, in system 1B, a high level signal (voltage value Vm x 1 second)
or 1/148 seconds, the total electric power E1 is: El = 1. /2VII, 2x1/ 148 (29) When comparing with System 2, it should be noted that VIIl corresponds to the peak voltage value of all time slots 81 to 5148 in one frame of System 2. (However, as will be explained later, this will lead to a severe situation).

On the other hand, in system 2, a high level signal (voltage value yII
, xt/100 seconds) or 148 multiplexed signals, and how many are medium or low level signals? In other words, if V is the voltage of each time slot, the total electric energy E2 is: E2 = 1/2ΣV; In this case, it should be considered as around 15 dB instead of +9 dB, so finding the ratio r between equation (29) and equation (30), r=201og148-0.15=28 (dB)(3
1) From the above, the power of the modulated signal of system 1B within a certain period of 1/148 seconds is 2 compared to that of system 2G.
It was found that because the modulated wave was as high as 8 dB and had undergone a large modulation shift (modulation depth), there was a possibility of causing large interference to adjacent channels. but,
The above level difference of 28 dB is actually a multiple load gain of 2.
is equal to 1 dB. This is because if the comparison time and peak power amount are selected accurately, there will be a difference in multiload gain between 1 audio channel and 148 audio channels (24.6 channels in FDM conversion).

The above results can be further explained as follows. System 1B stores high-level audio signals for as long as 1 second, compresses the time, and sends them out as FM signals within 1/148 seconds, whereas System 2 stores high-level audio signals for only 171,000 seconds. It is stored in memory and compressed to 1/148. Then, similar processing is performed for the other 137 time slots within one frame, and various signals such as high-level audio, medium-level audio, or mostly silent audio are detected within 171,000 seconds. They are arranged in chronological order and sent out into space as FM signals.

As a result, the modulation signal level within one frame of system 2, and ultimately the modulation deviation amount (modulation depth Md) of system I
This means that the multiplexing gain (21 dB) is much lower than that of B. Therefore, interference to adjacent channels is also reduced by the amount of modulation shift.

Of course, it is clear that the modulation deviation amount of system 1B does not cause interference if the audio signal within one time slot is silent, but in system 2, because the time of one frame is shortened, While the average power within one frame is low, in the system 1B, this average power is not constant and fluctuates greatly.

I guess this is the problem.

(7) Countermeasures against peak dehesion of TCM-FM signals As explained above, it has become clear that transmitting power can be significantly reduced by using the multiplexed gain of the TCM system to increase the amount of modulation deviation. Therefore, we will explain countermeasures against peak dehydration. Although the time stochastic [J is small, it cannot be denied that even in the TCM-FM signal (previously) system 2), peak deviation occurs, which may cause adjacent channel interference.

First, as used in mobile communications in general, the input stage of the modulation circuit of a radio is equipped with an IDC (In5tantane).
A circuit for instantaneous modulation deviation amount suppression may be provided to provide a function to limit the modulation depth to a certain value or less. If this is used, the modulator output will be regardless of the TCM telephone signal condition.
This means that the maximum modulation shift is kept below a certain level.

In addition, as another countermeasure, microwave analog (FDM)
-1800CH, etc.) experience is helpful. In the microwave analog system, a bandpass filter is inserted into the transmitter output (antenna feed line input) to remove high-order sidebands of the F~1 signal and prevent interference with adjacent channels. A method completely similar to this may be adopted for D-C to 1-FM.

Furthermore, by using a compander to be described below, it is possible to expect effects equivalent to those described above.

(8) Regarding the use of a compander A compander is a compressor (compressor) and an expander (
This is a general term for combinations of expanders (expanders), and those used in analog voice communications operate in response to the envelope level of the voice, and are also called syllabic companders, which are currently widely used.

In general, in mobile communications, the reception level fluctuates greatly by 20 to 3 QdB due to fading, so the reception S/N (
Compared to the case without fading, the signal-to-noise ratio (signal-to-noise ratio) deteriorates significantly compared to the case without fading, and even when the reception level is quite high, various noises enter during a call, resulting in inaudible interference. Noise includes thermal noise, click noise, and random FMiffl sound, but click noise is especially noticeable when there is no call, and even if the line is designed to ensure sufficient single-tone intelligibility, it will greatly deteriorate subjective evaluation. . Compander increases the audio level in the wireless section during calls,
It has the effect of improving S7.' average and greatly suppressing the noise generated in the wireless system when there is no call, and is a very effective means as a technology for improving call quality in mobile communications.

In the present invention, a compander similar to the above is used to compress the peak value of the CM signal. The specific system configuration and operation will be explained below.

FIG. 1D and FIG. 1F each illustrate a mobile radio device 100.
This is an embodiment in which a compander is applied to the radio base station 30. In FIG. 1F, the compressor 71 is inserted between the signal allocation circuit group 52 and the wireless transmission path 32,
Here, the amplitude of the time compression multiplexed signal is compressed. An example of the compression characteristics experienced at this time is shown in FIG. 9A.

The compressor characteristics in FIG. 9A are for the input signal n (n
Since it gives an output of l, , /2, 1. /2 compression. V, that is, if the input level changes by 10 dB, the output level will change by 5 dB. Therefore, the distribution of amplitude characteristics of the audio signal is 1/'2 in decibels. However, when entering the wireless transmission circuit 32,
Compared to the case where the compressor 71 does not compress the signal, the amplitude distribution of the signal applied to the wireless transmission circuit 32 is 1 decibel.
/2.

Now, assume that an amplitude-compressed signal is transmitted from the radio base station 30B from the antenna and received by the mobile radio device 100B. As shown in FIG. 1D, the configuration of the mobile radio device 100B is such that the time-compressed audio signal of the received signal is input to an extract bank (decompressor) 172 inserted at the output side of the receiving section 137. Ru. This input signal is the 9th A
It will undergo transformation according to the expander characteristics shown in the figure.

That is, when the input level changes by 5 dB, the output level changes by 10 dB. As a result, as shown in the center of FIG. 9A, the overall characteristic is that for every 1 dB voice input change on the transmitting side, the receiving side telephone input change is also 1 dB, and the original signal is faithfully reproduced.

The above explanation is based on the radio base station 30B transmitting, mobile radio device 1
Although we have explained the case where 00B is received, mobile radio 100
The situation is exactly the same when transmitting from the radio base station 30B and receiving from the radio base station 30B. Thus up.

In the case of downlink calls, the user will hardly notice the presence of the compander, and it is expected that the call quality will be improved.

Companders with 1/2 compression (expansion) characteristics are more likely to have arbitrary ratio (a/□b) compression (expansion) characteristics, so Figure 1D and Even in the 1F diagram, depending on the case, it may be appropriate to use a compander other than the 1/2 characteristic, and the decision must be made while considering the system conditions.

Plus companders (compressors and expanders)
With respect to the multiple load gain of the CM signal,
It is possible to increase the profit 1q beyond the value obtained from equation (25), which is isometrically converted to the FDM multiplex number. However, in this case, it is necessary to pay some compensation, such as generation of distortion noise and instability of system operation, but in practice, this is minimized to increase the gain. Take system 1B in FIG. 7 as an example.

As explained above, in system 1B, the multiple load gain q′
is calculated from equation (25).As explained above, increasing the modulation shift of the modulator inevitably causes interference with adjacent channels. However, considering that a compressor with a high compression ratio is inserted into the modulator input and the peak value is removed by an instantaneous modulation deviation suppression circuit,
Due to the function of these circuits, the modulated wave is
For example, the modulation deviation (modulation depth Md) is 20d.
Even if B is increased, modulated waves without interference can be obtained in adjacent channels. This will be explained further.

The characteristics of the compressors C1 and C8 used in the CM systems 1A, 1 and 1B of FIG. 7, respectively, are shown in FIG. 9B. In FIG. 9B, the horizontal axis shows the input level of the two types of compressors CA or CB, and the vertical axis shows the output levels of CA and CB. Point A and point B on the horizontal axis indicate the average input level for one frame time of TCM systems 1A and 1B, respectively. Here, Mv is related to one frame in which points A, A, B, and B both have a high level of power, even though it is the average power for one frame time.

The compression ratio of the compressor in this embodiment is OA or 1/2 (input 2 dB change (output change 1 dB>), C8 or 1.
'10 (output change 1 dB for input 1 Q dB change).

Now, in the Ding CM system 1A, the input level is point A, and the compression ratio is 1. Assuming that a compressor CA of /2 is used,
When the peak value is 1Qd&higher than point A, the output level of compressor CA will be suppressed to 5 dB, but the output of compressor CA is equipped with an instantaneous modulation deviation suppression circuit. Output level is 3d
The signal is completely suppressed at point B, and a level of 3 dB or more is not applied to the modulator.

On the other hand, since the Ding CM System 1B has a long frame time,
Normally, it would be best to operate at point A because only a multiplex gain can be obtained like in TCM system 2, but since we want to reduce the transmitter's transmission power (approximately 20d13), we set the input level of compressor C8 to point A. This is raised to point B, which is 2Qd13 higher than that of B, and is set as the operating reference level.Then, the output level becomes 2dB higher than the input with the average power of one frame containing a high signal level, and the system 1B signal The peak value of
Since it is a QdB raise, it becomes 10 dB higher than point B, but the increase is only 1 dB due to compressor C8, and the output level becomes 3 dB. Therefore, the output level will be approximately equal to the output level of the system 1A, and no interference will be caused.

The above explanation of the operation was based on the average power of one frame with a high signal level for both system 1A and system 1B, so considering the average power of the CM signal over a long period of time, for example 10 seconds or 30 seconds, the compressor operation The reference level is considerably (approximately 20 dB to the left) from point A and point B. However, even for these low levels, the output of compressor C8 is at a high level compared to CA, so
Since a high-level signal is always applied to the modulator, a modulated wave with a large modulation shift is obtained, making it possible to gradually reduce the transmission power.

In order to reduce the influence of distortion noise on communication as much as possible while obtaining the above advantages, it is necessary to use a compressor with good characteristics and a high compression ratio.

Although the compander described above has exactly the same principle of operation as the currently commercially available audio compander, the required characteristics are different from those for audio (0,3 to 3k)lz).
Characteristics that allow operation at extremely high frequencies are required. This is because, as is clear from the above explanation,
The frequency range of the input signal to the compressors 71, 171 is extremely high frequency and wide because the audio is time compressed and multiplexed. For example, example 1
In case of 48 multiplex, the frequency range is 0.3kH2~3k
148 times Hz, 44.4kHz2 to 444kHz7, and as explained above, the signal level fluctuation must be assumed to be a value equivalent to the standard deviation σ of the amplitude of one audio channel centered on the average power value. No. In a system that increases the modulation deviation by increasing the apparent multi-load gain, the level fluctuation of the input signal is large, so characteristics that operate well over a wide dynamic range are required. Also,
The expanders 72, 172 used in the receiver also need to have performance that satisfies the above operating conditions. In the past, companders used in TCM-FM systems required more advanced technology than companders for cordless telephones and car telephones that are currently widely used.

The above is a so-called ideal compander required for CM-FM, but below, an example will be described in which a practical or economical compander is applied, although it is not perfect theoretically. For this purpose, consider the application of currently commercially available companders to the present invention.

1E and 1G show the mobile radio device 10 in this case.
The configuration of 0C and radio base station 30C is shown. In the configuration of FIG. 1G, a combustor group (signal compression circuit group) 71 is inserted between the signal processing section 31 and the signal speed conversion circuit group 51, and the audio signal is compressed here. For example, in compressor 711, the audio signal undergoes level conversion as shown in the compressor characteristics of FIG. 9A. Once upon a time,
n audio signals pass through a group of compressors 71, and a group of signal speed conversion circuits 51. When entering the wireless transmission circuit 32 via the signal allocation circuit group 52, compared to a signal that is applied to the wireless transmission circuit 32 without passing through the compressor group 71,
The amplitude distribution is 1/2 in decibels.

Now, assume that the above signal is transmitted from the radio base station 30C via the antenna and received by the mobile radio device 100C. As shown in FIG. 1E, the configuration of the mobile radio 100C is such that the received signal is input to the expander 172 after passing through the speed restoration circuit 138. Here, the received signal is subjected to conversion according to the expander characteristics shown in FIG. 9A. As a result, as mentioned above, the original signal is faithfully reproduced.

The above explanation is based on the transmission from the radio base station 30C and the mobile radio 1.
Although we have explained the case where 00C is received, mobile radio 100
It is exactly the same when transmitting from C and receiving from radio base station 30C. Thus up.

In the case of downlink calls, the user will not feel the presence of the compander at all, and it is expected that the call quality will be improved.

(9) Regarding card time between time slots used in TCM-FM signals The advantages and disadvantages of an example in which a cart time is provided between time slots of a TCM signal according to the present invention will be explained.

The TCM signal described above does not necessarily require a cart time between pulse trains like a digital signal. However, in order to eliminate the influence of timing deviations of synchronization signals and delayed waves due to multiple waves on radio wave propagation, card times may be provided between time slots. The actual value of cart time varies depending on the system to be applied, but for example, it is 0.1 to 0.5 μsec for an indoor mobile phone system, and 5 μsec for a car phone.
~10 μsec is appropriate.

In a system with a cart time, if the frame length is constant, the width of the slot time decreases by the size of the cart time, so the compression ratio of the original signal must be increased. The highest frequency of becomes higher. In the cordless phone example above, the time slot is 1 m5eC ÷ 500 = 2 μsec, which is 10%
In other words, if we take a card time of 0.2 μsec,
The time slot will be 1.8 μsec. Also, the highest frequency is 3KH2X 500 without cart time.
= 1.5MHz xlO,/9 = 1.67 MHz if we take 10% cart time from 1.5MH2
becomes. However, the required bandwidth will be as wide as the required bandwidth.
This means that the frequency effective utilization rate will decrease by 11%.

Next, we apply multiload gain to the amplifier design. In this case, the level of the multiplexed audio converted into TCM may be considered to be lower than the level conventionally considered by the amount of the multiplexed audio gain. Therefore, even if the amplification factor can be increased by that amount, or the output level can be made higher than the conventional multiplex gain, the distortion factor, etc. will remain at the conventionally assumed value.

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. In other words, if applied to a mixer circuit, even if the rated output is increased by the amount of the multiplex gain, it is possible to operate in the conventionally assumed operating state. When applied to a wireless transmitter, this has the following benefits. For example, the first
Inserting a power amplifier into the output of the transmission mixer 133 in Figure B is often used to increase the reach of radio waves. In this case, by introducing the multiple load gain, it is possible to increase the transmission output level by an amount indicated by the multiple load gain than the conventionally assumed level. Alternatively, if the same transmission level as the conventional one is sufficient, the amplifier's rated output 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 the transmission mixer but also to all of the resistors, capacitors, inductances, etc.

[Effects of the Invention] As is clear from the above explanation, as a result of quantitatively clarifying the multiplexing gain of a time-division time compression multiplexed signal, which had not been clearly shown in the past, using system parameters, for example, , the depth (deviation) of angle modulation is
It is possible to suppress the influence on other wireless channels within the conventional design value even if the transmission is made deeper by the amount of tsubo, and the transmission output level per wireless channel is gradually reduced compared to the conventional system. It became possible.

Also, if we increase the modulation deviation by the amount of multiload gain,
When the peak value of a signal is high and there is a possibility of causing interference to adjacent channels or radio channels using nearby frequency bands, it is better to use an amplifier, which can be prevented by using a compander. Since it has become possible to create rational and economical designs ranging from the design of passive elements to the method of determining the ratings of passive elements, the effects on communication systems, especially wireless systems, are extremely large.

[Brief explanation of drawings]

Figure 1A is a conceptual block diagram showing the concept of the system of the present invention, Figure 1B is a circuit diagram of a mobile radio used in the system of the present invention, and Figure 1C is a wireless radio used in the system of the present invention. The circuit configuration diagram of the base station, FIG. 1D and FIG. 1E are other circuit configuration diagrams of the mobile radio device used in the system of the present invention, FIG.
Figure 2A is a schematic circuit diagram of a wireless base station used in the system of the present invention, Figure 2A is a time slot structure diagram for explaining the time slots used in the system of the present invention, and Figure 2B is a diagram of the time slot structure for explaining the time slots used in the system of the present invention. Figures 3A and 3B are spectrum diagrams showing the spectrum of speech signals and control signals; Figure 3C is a circuit configuration diagram for multiplexing voice signals and data signals; Figure 4A is a diagram showing radio signal waveforms of time slots; 4B and 4B are flow charts showing the flow of operation of the system according to the present invention, FIG. 5 is a spectrum diagram of a frequency division multiplexed signal, and FIG. 6A is an amplitude diagram showing changes in amplitude of a time division time compression multiplexed signal. , Figures 6B, 6C, and 6D are sampling diagrams showing how the time division time compression multiplexed signal is sampled, Figure 7 is a specification diagram showing the specifications of various systems, and Figure 8 is a diagram showing the specifications of various systems. A time slot structure diagram to explain the time slots used by the system shown in the figure, Figures 9A and 9B are characteristic diagrams showing the input/output characteristics of the compander, and Figure 10 is a time division time compression diagram. Figures 11 and 12 are diagrams showing the relationship between the multiple load gain 1 tsubo of a multiplexed signal and the number of voice signals to be multiplexed, and the relationship between the multiple load gain 1 n of a frequency division multiplexed signal and the number of communication paths cited from a known document. It is a multiple 9 load gain diagram showing the relationship. 10... Telephone network 20... Gateway switchboard 22
-1 to 22-n...Communication signal 30...Radio base station 31...Signal processing section 32...Wireless transmitting circuit 35...Radio receiving circuit 3
8... 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 section 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 71...Compressor (details) 72...Expander (group) 91...Digital encoding circuit 92...Multiple conversion circuit 100 .100-1 to 100-n-...Mobile radio device 10
1...Telephone unit 120...Reference crystal generator '121-1.121-2...Synthesizer 122-
1.122-2...Switch 123...Transmission/reception intermittent controller 131...Speed conversion circuit 132...Wireless transmission circuit 133...Transmission mixer 1
34... Transmission section 135... Radio receiving circuit 1
36... Reception mixer 137... Receiving section 138.
... Speed restoration circuit 141 ... Clock regenerator. 171...Compressor 172...Expander.

Claims (1)

[Claims] 1. Each radio base means (30) each covering a plurality of zones to constitute a service area, and a means 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 temporally compressed delimited signals in time slots having a frame structure; In the mobile communication method used,
The level of the radio signal used for communication between the radio base means and the mobile radio means is determined based on the multiple load gain obtained by the time-compressed segmented signals, and the signal to be transmitted is transmitted through a compressor. Means (71, 171
) is used to compress the amplitude distribution to a desired range before transmission, and the received signal is restored to its original amplitude distribution by an expander means (72, 172). 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 radio base means and the mobile radio means each determine their respective transmission output levels based on the multiple load gain obtained by the temporally compressed and segmented signals, and adjust the amplitude distribution of the signals to be transmitted within a desired range. compressor means (71, 171) for compressing and transmitting the received signal; and expander means (72, 172) for restoring the received signal to the amplitude distribution before compression.
A time division communication system for mobile communication comprising: