UA73724C2 - Method for transmitting information and a system for the realization of the method - Google Patents

Abstract

Disclosed is a method for transmitting information and a system therefor which is particularly suitable for digital transmission. At least one information signal consisting of a reference frequency channel and at least one information frequency channel is generated, whereby the reference frequency channel and the information frequency channel respectively form discrete states in order to provide a bit pattern. This enables a signal to be transmitted over a distance of several kilometers, under water for example. A suitable evaluation system is also described.

Description

Description of the invention

This invention relates to the development of a method of information transmission and a system intended for this purpose. 2 In many branches of technology, waves are used to transmit information. At the same time, we may be talking, for example, about electromagnetic or acoustic waves that, being emitted by the transmitter, reach the receiving device, propagating along a special waveguide or freely in a certain environment. If the receiver and transmitter agree, for example on the frequency or on the selected frequency range, the connection is considered established. With the help of such a connection, information can be transmitted in different ways. 70 For this, the transmitted information (in the form of speech, text, a series of numbers, graphs or other data) can be transformed or encoded in order to be emitted by the transmitter in the form of wave signals. The receiver receives signals, transforms them into their original form, that is, demodulates them, and gives them at the output in the appropriate form.

Depending on the form of information modulation, there are analog and digital methods of transmitting 12 information.

With the analog method of information transmission, the transmitted values are represented in a continuous non-step spectrum of physical states. This usually occurs in the form of modulation of the amplitude, frequency, and/or phase of the carrier waves. On the contrary, with the digital method of communication, they are limited to only a few discrete states.

In today's practice, there are still no restrictions on the speed of information transmission in the case of the use of electromagnetic waves, since the frequencies of the corresponding carrier waves are very high and therefore various discrete states can be realized in exceptionally short time intervals.

In some mediums of information transmission, for example in water, communication using electromagnetic waves is only conditionally possible, since these waves propagate there only over very short distances. In such a medium, however, it is possible to transmit information by means of sound waves, which can propagate over (3) much greater distances. Sound waves can be modulated in the same way as described above. These waves, however, are mechanical pressure waves, different from electromagnetic in nature propagation and, in addition, significantly lower frequencies, which ultimately affects the speed of information transmission. Regarding the nature of the propagation of mechanical waves, there is, for example, a significant dependence of the speed of sound on certain environmental conditions.

Numerous problems encountered in the acoustic method of data transmission are briefly explained in the next section using the example of underwater sound signal transmission. When propagating in the water environment, the sound waves emitted by the transmitting hydrophone can be reflected or deflected, for example, from the surface of the water and/or, depending on the depth, from the bottom, various objects, air bubbles, suspension, as well as from layered inhomogeneities of the water channel Different components of sound waves enter the receiving device with different amplitude and phase, depending on the length of the propagation beam, angular relationships and acoustic properties of the reflecting boundaries. The useful signal at the reception point can, as a result of interference, be unpredictably strengthened, weakened, distorted or completely disappear; reception, "as a result of the so-called echo, may be disturbed. C 70 For a more complete explanation of the problem, let's first consider a simple case when a very short signal of a certain frequency is sent, the so-called SMUR signal (Sophiipiosiz UUame Ryivze). Then, under the conditions mentioned above, not only one single signal can be received, but a whole group of time-shifted individual pulses of different strengths. This effect is determined by the transfer function of the "channel feedback" type. For the described case, it is possible to distinguish when receiving individual pulses from each other and, for example, to choose the strongest pulse as "own signal", while other pulses can be described as "interference" and accordingly. way, as obstacles, processed . A similar possibility of separating multi-beam pulses from one another is usually not possible for longer wave packets. This is explained by the fact that the receiver registers a total (or composite) signal, which, although it contains the former value of the frequency (as well as the radiated signal), represents, however, the superimposition of interference signals with different phases and amplitudes, leading to unpredictable jumps phases and amplitudes in the total signal. This negative effect, which complicates or even makes signal processing impossible, is called "intersymbol interference". IN

In the case of relative movements of the receiver and transmitter, an additional problem of Doppler frequency shifts arises.

A number of the aforementioned difficulties make underwater communication, such as ultrasonic between divers and/or an underwater vehicle, extremely difficult. Remote control of underwater devices is also difficult.

GF) Until now, the analog communication used is only conventionally practical. It was and still is commonly used for voice communication, using the properties of human hearing to recognize familiar words and meaningful context even with very noisy reception. With appropriate training and an agreement to use a limited list of teachers, some improvement in speech recognition can be achieved. This approach cannot, however, be used to transmit, for example, computer data or other information in an algorithmic way. For this reason, the search for a suitable method of digital exchange is also underway in the field of acoustic communication.

Modern technical digital systems, especially for underwater applications, are based mainly on the transmission of tone signals of constant pitch, which lie in a more or less narrow frequency band. In order to achieve the greatest distance of exchange and to exclude the loss of information in acoustically "dead" zones, in some systems, high-energy signals are emitted in a wide frequency band. Regardless of whether the signals are transmitted in a wide or narrow band, the modulation method of the type of successive "clicks" (amplitude-modulated signals) allows you to achieve only a very limited exchange rate, which makes it difficult or impossible to transmit large amounts of information, for example, underwater camera frames etc. In addition to relatively high energy consumption, which also means "acoustic pollution of the environment", the systems known today, to some extent inflexible, have large Doppler shifts.

In addition to the distortions and losses technically caused by transmission, there are also significant difficulties with processing the information contained in complex signals in such a way as to eliminate or weaken (when receiving) various 7/0 distortions and to reconstruct the signal parameters used for information coding. Among the methods of communication, there is no approach so far that allows to sufficiently and optimally solve the whole set of the mentioned problems.

The task of the present invention is to develop a method and a corresponding system for data transmission, which would allow communication over long distances with a higher speed of exchange.

Next, a method and system for transmitting information that is resistant to the above-described distortions and can be adapted to various communication conditions is developed.

It is especially important to develop a method and a corresponding system that allow for the effective selection and subsequent analysis of those components that have suffered the smallest losses in the channel, and which are isolated from the composition of numerous multi-beam components that represent the phenomenon of intersymbol interference in their totality.

Next, a method and a corresponding system for signal processing are developed, which allows, in the context of the current invention, to also carry out, if possible, full compensation of Doppler shifts.

By improving the quality of signal processing, in the future, it is expected to create prerequisites for a significant increase in the speed of exchange and, if necessary, also the range of exchange in difficult conditions of communication, such as, for example, between moving underwater objects.

These tasks are solved methodically using the properties presented in clause 1 of the formula i) of the invention and technically using the properties presented in clause 31 of the specified formula.

According to the application, an information signal is generated, consisting in the minimum version of two components - at least one reference component sent in the reference frequency channel, and as "E zo Minimum of one information component transmitted in the information frequency channel - so that several frequency channels or frequency components are available. With their simultaneous use, more units of information can be transmitted per unit of time. In the future, both the reference M channel and the information channel provide for use discrete states that form a bit pattern.

A significant difference to radio technology is the fact that it does not use a high-frequency carrier wave, which is modulated by a low-frequency wave, an information signal is formed here, which is involved in the transmission of information, which is a wave consisting of an overlay (sum) reference channel and at least one information channel.

To form a bit pattern, in the simplest case, frequencies (or tones) of information channels can be turned on or off, and the presence or absence of the corresponding frequency components of the "signal are considered binary digital states of COM" / "ORP"), that is, 1 or 0. In this way, it can to be transmitted one bit of information on each information channel.The components of the signal form together

And a bit pattern in which information can be encoded in any way. and?" In this simplest case, the "OM" states, other signal parameters may also vary so that additional digital states may become distinct.

Further useful forms of performing the tasks of the present invention are presented in the dependent clauses. -And According to point 2, the simplest temporal sequence of the bit pattern is formed.

The third point contains one of the useful variants of the invention, in which the frequency channels form a harmonic series. - And in the event that, in accordance with point 4, the information signal is formed in the form of the sum of the reference channel, which is the main tone (main wave) and, at least, one information channel, which are in a harmonic relationship to each other (for example, the information channel is an overtone of the fundamental tone), or i" also, if all information channels lie in harmonic ratios to the fundamental tone (fundamental wave), then the harmonic series or consonant system consists of separate frequencies (tones) or signal components.

The peculiarity of the system disclosed in this communication is that the main tone, which has the lowest frequency and the largest propagation distance, can be emitted during the communication session continuously, thus creating a seemingly continuous bridge between the emitter and the receiver. The reference channel formed in the form of the main (F, tone) serves in this case not actually for data transmission, but as a constant support (dense tone) in signal processing for the coordination of other information channels and, if necessary - as will be presented later - to match the relative values of the phase, and it is also possible as an energy "feeder" in the case of using non-linear effects to increase the transmission range of composite signals (the entire frequency system).

At this point it should be noted that instead of the lowest frequency, any other tone of the given frequency spectrum can be used as a reference channel (or main tone), in case it may become important for certain reasons of environmental protection or for a certain technical application.

By assigning the information frequency channels a certain distance from the reference channel 65, it is ensured that a receiving device, which knows the appropriate distances or proportionality factors, only needs to find the reference channel formed in the form of a fundamental tone, in order to recognize on this basis all other active information channels and be able to coordinate them promptly. This process of coordination allows such automation, accordingly, the system, without incurring significant additional costs, can be built for the most diverse communication conditions. Automatic recognition of the main tone and the corresponding adaptive matching of information channels on the part of the receiver creates significant advantages, especially for communication between moving objects, since, for example, problems caused by Doppler shifts, present in other methods, disappear here, for example, if used system of harmonic frequency channels.

In the event that, according to the fifth point of the formula, the frequency of the reference channel in the process of data transmission 7/0 Changes over time, not only a permanent extension of the adaptive system is formed (on this basis) to compensate for natural frequency shifts (Doppler effects, etc. ). It is much more important that the frequency spectrum is changed in a completely systematic manner on the part of the emitter, without creating a danger of loss of contact for the receiver.

In the event that the frequency change of the reference channel is continuous or stepwise, as stated in clause 6, then one or more values of the frequency change gradients may be provided for use. This approach is hereinafter referred to as the frequency gradient method (FGM). With this method, it is achieved that, for example, reflections or interference signals can be eliminated. The variation of reference and information components based on the method of frequency gradients will be referred to as the method of variable multi-channel transmission (VMT).

The case when the variation of the components is carried out proportionally is called the pREOM or pUMT method, and when the variation of the components is parallel, we call it the rAOM or raUMtT method.

With the help of the computer method, it becomes possible to perform a more accurate and reliable signal analysis than with the help of previous methods, especially those that use frequency channels with time-invariant frequencies. high school

Due to the fact that in this case the operating frequencies of the information frequency channels are constantly changing over time, all signal components arriving at the current moment of time to the receiver via different paths have different (instantaneous) and) frequencies. On the basis of these frequency differences, the actual informational frequency channels can be separated from the interfering components existing in each case; therefore, intersymbol interactions can be largely or even completely eliminated, while on the receiver side, a much clearer "E" can be restored from the display of the information signal emitted by the transmitting device.

Since in the computer method the frequency of the reference frequency channel and, in addition, according to the given ratio, the frequency of the information frequency channels can vary synchronously and according to almost any dependence, then both the method and the system that are claimed are very flexible. With the help of deliberately introduced frequency shifts, it becomes possible to avoid mutual overlap of several transmission systems and reduce the probability of unwanted eavesdropping. Cha

If, in addition to the frequencies of the reference frequency channel and the information frequency channel, other signal parameters are also involved to create a bit pattern, it becomes possible to complicate the coding in a simple way and, accordingly, to increase the data transfer rate.

If the amplitude modulation of the information signal is carried out according to point 7, then in certain nodes of oscillations "amplitudes (which are involved for such modulation) can be set moments of time in which, for example, individual c information frequency channels can instantly change their information parameters without causing interference . in the information signal at the points of such instantaneous changes. With this, the quality of information transmission can also be improved.

With the help of the measures outlined in point 11, an increase in data transfer speed can be achieved.

On the basis of the achieved high quality of reception (due to the use of the EOM method), in combination with -And the above-described method of turning on/off individual information components (creating binary states "OM""OGE") or instead of this method, information can be encoded using more of subtle changes in specified parameters or combinations of parameters in the signal. Since in the received signal, in addition to the frequencies, there are also -I amplitudes and phase angles of the signal components, which have a more defined correspondence to the generated values in the transmitted signal, then almost all parameters can be involved for coding. This can happen where, for example, with the help of step changes. A significant advantage of this method is that internal relations in the frequency system can be used for coding. With the help of such relations, it is achieved that bit or pattern symbols can be identified already on the basis of one or two received signal clocks, and without the need to involve an additional reference to an external reference value.

So, for example, phase angles can be specified in a given time cycle as the current ratio between

F) corresponding parameters of the information component and reference channel. ka This coding method is called the relative phase angle method (KRUUM). Prehistory no longer plays any role in this method, external time loses its significance for signal processing. Instead of it, the internal (relative) system time is gaining strength, which can be calculated, for example, based on the instantaneous values of the cyclic time, which in turn depends on the current value of the frequency. Relative phase angles can be determined in a simple way, when, for example, in the process of processing, all signal components, that is, reference and information channels, are first normalized to one universal period duration. This principle is explained as follows. In the theory of signal processing, there are many approaches that use projection and transformation 65 of the signal, which can be applied in determining the relative phase angle. At the same time, the user is provided with many opportunities for their practical implementation. For the proposed method, however, it is essential that as a result of the use of EOM and especially pREOM, a number of effects of interference can be eliminated; at the same time, differential phase angles can be determined with great accuracy, which can be used for finer discretization, that is, for distinguishing a greater number of digital states, and for further increasing the speed of information transmission.

The next option is that the information is encoded indirectly as the ratio of the phase angles of the corresponding components and the reference channel (the main tone), that is, as a "vertical" internal relationship, but as the difference between the phase angles in the current and previous clock of the same component, as, say, a "horizontal" internal relation. This approach is called the differential-phase method (DFM). In 7/0, each first beat of the corresponding component serves as a horizontal reference. In difficult communication conditions, it can be very useful to use a combination of the CROM method and the method proposed in point 9.

On the other hand, in the case of simple communication conditions, it may be sufficient to use only the horizontal internal relation. In this case, the reference frequency channel can serve as an information frequency channel. Next, it should be noted that both with KRUUM and with KROM, the fact of the absence of one of the 7/5 components of the signal (or the fact of not exceeding the specified amplitude threshold) can be used to encode an additional digital state.

If, in accordance with point 12, the number of information channels varies depending on the properties of the transmission channel, then, especially with a decrease in the distance between the transmitter and the receiver, additional, usually high-frequency (or lying in the gaps between the used) th channels, for example, consonant frequencies, can be used ; and vice versa, with very long communication distances, mainly low frequencies are used. This measure achieves optimal use of the nature of the propagation of wave signals, which is very important, especially when using audio signals. In this way, maximum communication speeds and/or communication ranges can be achieved, for example, in underwater channels. Of course, it is possible to use this method flexibly, so that the settings used for certain types of communication conditions can be taken as the basic standard, provided that the given operating mode is preserved. and)

In addition to specific states or proportions of signal parameters, this approach can also be used to encode information using instantaneous values or certain dynamic characteristics of the signal. If, as in point 13, individual information channels are specified in a wider frequency band but without overlapping, then the possibility of information coding using constant (non-stepping) phase shifts in the corresponding information component appears. This measure is defined as the M phase gradient method or the phase velocity method (VMF). The distance of the information components from the reference channel is determined then, in a typical case, as the distance between the corresponding lines of average values and)

From the frequency of the corresponding channels. When transmitting data, the frequencies of individual information frequency channels (for each y-time cycle) can change slightly and smoothly (of course, by less than 0.595 of the actual expected frequency value), while a continuous gradual (or accelerated) phase shift of the current component relative to the phase is carried out reference channel or main tone. The receiving device distinguishes not only the fact of the transmission of components of a certain frequency on a given clock, but also determines (in the case of the existence of a component) "the value of the relative phase angle and/or the characteristic parameter describing the function of the phase change in c with c depending on the actual cyclic time that is read from the reference channel (or main tone). Apart from their own

And values describing states or proportions can also be used for encoding smooth changes in time of such states or proportions. At the same time, many different possibilities of variation and combination are created, which allow to increase the speed of information transmission, to improve the adaptability of the system to different communication conditions, and also to optimize the composition of receiving and transmitting devices and their cost. -And To simplify the processing of the signal after its reception, the reference channel is separated from at least one component, as specified in point 16. o In accordance with point 17, with the help of pair processing of the corresponding information component of the signal -I together with the reference channel by the reference component (or one of the most suitable reference or reference 5r Components) the possibility of compensating Doppler effects is achieved. As an additional result, this signal processing step can help in the preparation of frequency stabilization. In the case of using the radioM method, this step can lead to the formation of stable, i.e., time-invariant, intermediate frequencies.

The superstructure of the method according to clause 18 ensures the translation of the signal components into stable (time-invariant) intermediate frequencies, which allows further useful processing. One of the advantages achieved by the implementation of clause 18 is, for example, that stable intermediate frequencies can be formed in the optimal frequency window for subsequent filtering according to clause 20, which makes, in addition,

F) it is possible to use special narrow-band filters. As an alternative to the approaches defined in points 16 and 18, when using REOM or rUMT, it is possible to generate stable intermediate frequencies only by mixing the signal received in the current cycle with the signal received in the previous cycle, and without the need for preliminary separation signal component and application of heterodyne frequency. This variant of signal processing is proposed by application Mo 19, especially for use in combination with the differential-phase method.

The superstructure of the method according to clause 20 has the task of isolating (for example, filtering) from the spectrum of now stable intermediate frequencies having a whole set of multi-beam components (or reflections), 65 only the most suitable signal components for each component and, at the same time, minimizing the influence of interference on them from other components. The latter implies the possibility of separation at this stage of processing from each other as well as the signal component, if this has not yet occurred (or has not occurred completely) in accordance with clause 16.

For this purpose, special filters can be used in the simplest case. At the same time, redundant components can be "turned off" (for example, filtered), that is, components that are not currently defined for processing. As a result of such processing, each information component of the signal is clearly expressed by one representative, on the basis of which the best recovery of information parameters (used for data encoding), such as amplitude and/or phase angles, becomes possible. This description is only about presenting the basic principle. It goes without saying that more complex methods of signal processing can also be applied here (from among the rich repertoire of known approaches), which, for example, in addition to identifying the components of the received signal, also deliver their parameters.

When improving the method of item 21, it is achieved that with the help of the approach proposed in this application, errors do not occur in signal processing.

As a result of the further superstructure of the method with the help of item 22, it is achieved that such multi-beam components (signal components) can be constantly identified for actual communication conditions, on the basis of /5 of which the parameters of the transmitted signal can be determined optimally, that is, as best as possible. As a rule, these are the most powerful components of the signal, which make it possible to achieve the best quality of signal processing. In the process of channel tuning, for example, the best filter settings can be determined for the most accurate filtering of the desired signal components and optimal suppression of interference from other multipath components and the possible influence of other side frequency bands. The latter can, in some cases, contribute to increasing the radius of signal reception and/or also increasing the speed of information transmission. The better and more reliably the received signals can be processed, the more possibilities can be available, the more fine discrimination of digital states or different combinations of parameters can be applied to information coding.

With the help of the current updating of filter adjustments, in accordance with point 23, optimal reception results can be achieved even in fast-moving communication conditions, and the advantage of using the named method is that it is not necessary to carry out tuning to the channel i) interruption of the actual transmission information

With the help of point 27, the optimization of Doppler compensation is achieved.

The method according to clause 28 is intended primarily for processing received signals under the influence of strong Doppler shifts, where each component of the signal is represented, however, in the main part by only one multi-beam component. with

Further useful forms of implementation of the proposed invention are the subject of other dependent clauses. M

Taking into account the drawings, various forms of implementation of the subject of the claimed invention are described in more detail below. at

Fig. 1 represents the construction of one of the options claimed in this patent, consisting of one reference frequency channel and three information frequency channels.

Fig. 2a represents an information signal composed of the components of FIG. 1 and subjected to amplitude modulation.

Fig. 26 represents a series of information signals divided into clocks.

Fig. C schematically represents one of the options for encoding information. "

Fig. 4 represents coding as in FIG. With, but in the version of the parallel computer method (ragoM). with c Fig. 5 represents the analysis of the signal at the moment of time Her. Signal made according to the principle of proportionality

ESM-method and composed of one reference and three information frequency channels (which are in harmonic relation to each other) is supplemented at the reception with one leading and one lagging interference component (multi-beam component).

Fig. 6 represents the basic principle for improving the analysis of the signal in relation to the interference components -I, respectively, Fig. 5 when using one reference and four information frequency channels.

Fig. 7 schematically represents one of the options for applying a stepped frequency shift with an additional change of information frequency channels within a clock, and in each case the first half of clock -I forms an additional horizontal reference for the KROM method.

In Fig. va schematically presents one of the coding options, however, creating only two stages of frequency change. i" Fig. 86 presents as an example the principle of pentacoding of information on one of the information frequency channels.

Fig. Za and 96 represent two different phase gradients formed on the signal constructed according to the principle of the two pROM method.

Fig. 10 presents different phase gradients that can be formed using the pROM method (top) and

F) pROM method (below). ka Fig. 11 represents the basis of construction of one of the claimed transmission systems.

Fig. 12 represents the development of the construction scheme of one of the claimed transmission systems using amplitude modulation.

Fig. 13 represents the basis of the construction of one of the claimed receiving devices according to its first embodiment.

Fig. 14 represents the development of the construction scheme of one of the claimed receiving devices with additional detection of phase angles according to its second embodiment. 65 Fig. 15 represents the analysis of the signal at time 3, the signal performed according to the parallel principle

ESM method and consisting of one reference and three information frequency channels (which are in harmonic relation to each other), supplemented at the reception with one leading and one lagging interference component (multi-beam component).

Fig. 16 schematically presents several examples of advantageous locations of frequency channels for various technical applications.

Fig. 17 schematically represents one of the main signal processing processes according to the claimed method.

Fig. 18 represents an example of frequency channels of a received signal, changing in time, formed using the pUMT method, consisting of one reference and three information frequency components under practically ideal communication conditions (a minimum of intersymbol interactions).

Fig. 19 represents the received signal according to FIG. 18 after transformation of the first information component of the signal to an intermediate frequency.

Fig. 20 is an example of the fact that due to the changing properties of a multi-beam channel, significant fluctuations in the levels of various components of a given received component can occur.

Fig. 21 represents an example (already shown in Fig. 20) after passing a narrowband filter.

Fig. 22 schematically represents one of the main processes of the claimed method, in which channel debugging is performed.

Fig. 23 schematically presents an overview of the most important signal processing steps in various useful embodiments of the claimed method.

Fig. 24 represents the basis of construction of one of the claimed signal processing systems according to its third embodiment.

Fig. 25 represents the basis of the construction of one of the claimed systems for carrying out debugging on the channel.

In Fig. 1 presents a version of the formation of the information signal IZ, which consists, for example, of one reference component, which is the reference channel VK, which in this case is made as the main tone of ST, and three information components, which are information frequency channels Ii, 12, IZ. Presented on i)

Fig. 1 information frequency channels are the harmonic overtones of NK, NK2O and NKZ of the main tone ST, the sum of which constitutes the information signal. From this figure, it can be concluded that the information on each information channel can be encoded by changing binary states such as channel "on" or "g" or "off", which corresponds, for example, to digital codes "1" or "0".

In Fig. 2a shows the amplitude modulation of the information signal IZ (taken from Fig. 1) in order to ensure a smooth (continuous) transition at the beginning and end of the clock for the case when the information signal M belongs to a certain information frequency channel changes in time.

This kind of change is presented in Fig. 26, and the value of the information signal changes from the clock me)

From the beat so that, for example, in sector I there is an information signal consisting of the sum of the fundamental tone, the second and third harmonic overtones (ZTANK2NKZ), which in the next beat (in sector II) will be transformed by turning off the second and third harmonic overtone to a signal consisting only of the fundamental tone, in order to form another information signal (corresponding to another coded bit pattern) representing the sum of the fundamental tone and the first overtone (see sector I) in the next beat. "

In this way, one bit per unit of time (cycle) can be transmitted on each of the information channels. As a result, a bit pattern is created from this, in which information can be encoded in any way.

In general, depending on the number of used information frequency channels of the applied coding system, a letter or any other symbol can be coded, for example.

It should be noted that when using 2, 4, 8, 12, 16 or another number of information channels, direct compatibility with various existing methods of electronic data processing can be achieved. -And in Fig. With is presented, for example, the word "OoIrpipSot" when it is transmitted in the generally accepted code of ASIA using four information channels. In this example, the system of frequency channels that makes up the information signal consists of one reference frequency channel (in the form of the fundamental tone) and four -I harmonic information channels (II, 12, ИЗ and 14, in the form of overtones of the fundamental tone), which vary in 5 hours using the proportional ECM method. The encoding in this example is done only by turning the overtones on/off. The vertical lines show bars that always have the same duration. Each bar creates a special bit pattern, defined as a symbol. Two symbols together form one letter. in the AZSIA code. The represented word is "boIrpipSot". In general, any other code can also be used to encode the transmitted information, which gives the user maximum freedom for his own programming, and also makes this method compatible with almost all other computer systems. As shown in Fig. 3, the reference frequency channel changes its frequency progressively (smoothly), moreover

F) the frequencies of the four information frequency channels (II, 12, III and 14) change in proportion to the frequency of the reference frequency channel.

In Fig. 4 presents the newly transmitted word "boIrpipSot" in the AZSIA code when using also four information channels, making the frequency shift of the reference frequency channel also progressively as in Fig. 3, where the frequency shifts of the information frequency channels (arranged initially in the harmonic ratio), however, unlike the case in Fig. C, are produced in parallel with the frequency shift of the reference frequency channel.

In Fig. 5 shows how a significantly more accurate and reliable signal analysis can be carried out when, for example, a translational frequency shift in the sense of EM is produced on the reference frequency channel. At the same time, 65 for the example shown in Fig. 5 and based on Fig. 3, three information frequency channels were chosen, on which, in addition to the actual signal frequencies, in each case, the receiver also registers one "leading frequency" and one "late frequency" as interference signals, and the corresponding time shift on each of the information frequency channels is identical To explain the basic principle, the bars were not indicated. The vertical cut line (at time /, .) shows that on the receiving side the frequencies of all information components differ from each other. It is especially important to note the fact that, based on these frequency differences, the actual signal frequencies now become separated from the interference frequencies, and thus the intersymbol interaction can be either completely eliminated or greatly attenuated. It is also important in this connection that the amplitudes and phase angles of the received and thus "purified" signal components have a "pure" reference (ratio) to the reference frequency channel. When applying the ECM method 70, special frequency filters can be used to separate the actual frequencies of the signal from the corresponding interference frequencies. In Fig. 5, it can be clearly seen that the distance between the signal frequencies and the corresponding interference frequencies increases with the increase in the gradient of the change (shift) of the frequency aa, that is, with the increase of the corresponding speed of the frequency change. Since, on the presented in Fig. 5 system, the frequencies of all information frequency channels change proportionally, the gradients of frequency change related to high-frequency information channels, 7/5 Increase their values; at the same time, the quality of separation of relevant frequencies of the useful signal from interference frequencies increases.

In Fig. 6, this effect (or functional mechanism) is presented schematically for a system consisting of one reference frequency channel and four information frequency channels, which in each case have two adjacent interference frequencies. Plotted in Fig. b dashed line symbolizes the characteristic, of course, of the applied filter. From the drawing, it can be clearly noted that even for a constant width of the filter window, an increasing quality of separation of components for high-frequency information channels is achieved. In comparison with existing methods, a significantly higher resolution is achieved in general. At the same time, it should be especially noted that according to this method, first of all, the most high-frequency information channels, the frequencies of which are suppressed the most in the propagation medium and, therefore, are received with the lowest energy, can be

Better separated from the signal with interference. Therefore, it is clear that, for example, at interference frequencies that are very close to the actual signal frequency, a correspondingly larger frequency gradient is selected, i.e., the frequency shift speed is increased, while for large distances between the useful frequency and the interference frequencies, it may be sufficient lower gradients. For adaptation of this kind, a pre-prepared set of frequency shift templates or a started adjustment of gradients in operational mode can be used. The latter can be simply feasible if the communication is used in two directions, that is, if the receiver can emit and the transmitter can also serve as a receiver. In this way, an analysis of the response of the channel can be carried out and the results of the analysis can be exchanged between, for example, the receiving and transmitting device or M, the appropriate selection of the bit pattern is also carried out, followed by the establishment of the optimal gradient of change (speed of shift) of the frequency. In general, such a gradient can degenerate to a zero value in the case, o

C5 if the communication conditions become accordingly favorable (stationary) and the obstacles become negligibly small. uh-

In this regard, it should be noted that in order to maximize the speed of information transmission, the change in the clock duration can be carried out in proportion to the change in the frequency of the reference channel, since no more than a certain number of oscillation periods is necessary to analyze the individual components contained in the composite signal. "

Figures 7, 11 and 86 present other possibilities that arise especially when applying a stepwise change in the frequency of the reference frequency channels. These possibilities represent particularly interesting alternatives to the EOM method, where the time shifts between the actual signal frequency and the interference frequencies represent rather large "" values, for example, in the millisecond range. In this case, a good separation of the actual signal frequencies from the interference frequencies can be achieved, even if all channels (on each beat) simultaneously and step by step

Change their frequency to a higher or lower value, keeping, however, this value constant during the -I clock itself. Also, in this case, it is useful to perform a frequency shift, if possible, so that the internal proportions in the signal are equally determined on all stages. This can be easier than anything achieved with proportional or parallel step shifts. This alternative is generally defined as the step-frequency -I method (ESM-method). Figures 7, 11 and 86 show how, for example, pentacoding of individual information channels is performed with the help of an additional phase state. In addition, to increase reliability, a reference signal is transmitted at the beginning of each cycle on all information frequency channels,

Ta" which is followed in the second part of the clock by a correspondingly coded signal. As shown in the comments to figures 7 and va, for example, five position points can determine 5 different discrete phase states, namely no signal (0) and four digital stages obtained, for example, by means of a CROM. In general, on each clock, for a 5Bv signal consisting of one reference frequency channel and three information frequency channels, 5!-125 combinations will be formed, which can be used for coding. iF) Fig. 86 represents an example of the principle of pentacoding of an information frequency channel. The points designated by 01-04 are signal components, the amplitude of which has exceeded the amplitude threshold Ao and four different phase angles (or angle sectors), which can be digitally separated from each other (a. State O5 demonstrates the case when the amplitude of the component signal does not exceed the threshold Ao.

A further way of encoding information is represented by the ROM method, and is explained in figures 3a, 96 and 10. Figures 3a and 96 represent the so-called method of proportional phase gradients or velocities (rPROM).

Fig. 10 additionally presents the relationship between the components for non-proportional phase gradients (pROM).

The main principle of the P(ozM-method) can be explained in the simplest way on the basis of the following initial 65 CO configuration.

Let this be a system in which information frequency channels continuously form a harmonic series, that is, frequency shifts are produced using the proportional ECM method. Take any information frequency channel, on which the corresponding frequency at a given interval (beat) is now formed not as an exact overtone of the fundamental tone, but slightly shifted from the "proper" overtone frequency down or up (of course, less than 0.5956 of such a proper value) , as shown in Fig. 10 in the top row. In the general case, a frequency shift occurs, which, however, is of such a small magnitude that this fact can hardly be recognized by the receiving device as frequency modulation and, therefore, cannot be interpreted as the value of a certain digital state. The frequency of such an information channel lies within the bandwidth of the corresponding narrow-band filter used for signal processing. Depending on whether this /o frequency is slightly above or below its "proper" value, as a result of its addition with the frequency of the fundamental tone

ST becomes apparent a continuous and smooth change of the phase angle of this frequency, as it is presented in figures 3a and 96. The phase of the frequency of the information frequency channel is either ahead of the main tone or, accordingly, behind it. In the corresponding beat, this creates a phase gradient, the direction of which can be recognized by the naked eye and can also be easily determined by one of the known algorithms. When calculating the 7/5 phase gradient in this case, a linear characteristic of the phase change relative to the instantaneous value of the duration of the main tone period will be formed. The prerequisite for such a linear characteristic is that with a constant frequency shift throughout the frequency system, the internal proportions set in the current clock, but easily changed, continue to be observed; - this also means that the relative frequency shifts in relation to the fundamental tone are unchanged. Fulfillment of this condition is defined as pROM-method, where P denotes the word 2o "rgoropiopai" (proportional). Observing the sequence of cycles, in the case of pROM, a uniform rotation of the phase of the information frequency channel to the left or right relative to the phase of the reference frequency channel will be formed.

This effect can be very advantageously used for encoding information, since, for example, when analyzing a rotating platform, the direction of the phase gradient is determined much more easily than the actual value of the phase shift. Mathematically, this means only determining the sign of the first derivative of the relative phase shift of the frequency of the information frequency channel relative to the frequency of the fundamental tone, or in other words i) determining whether the relative speed of the phase shift is a value greater than or less than zero.

This approach can be applied to each information channel individually on each clock. If this approach is used in combination with the previously described simple method of "turning on"/"turning off" the channel, then on a given clock for each information channel in relation to TZ, up to four different digital states are created: 1) no signal, 2) signal with positive phase gradient, C) a signal with a negative phase gradient and 4) a signal without a gradient, although in practice, the fourth state can hardly be used, tau Kk. in this case, four-position coding will be replaced by actually three-position phase gradient coding, which in some circumstances may still not be as reliable as two-position, since one of the digital states relies on a singular value of the phase velocity (MiYII). This problem depends, however, on the current communication conditions, because on the part of the transmitter, as a rule, there are no difficulties with generating gradients of any accuracy. Theoretically, the speed of information transfer could be doubled in the first case compared to a simple "on"/"off", and in the second option could increase by one third.

As an alternative to this, you can save the corresponding number of channels, and the frequency spectrum of the system "

Can be kept narrower, which provides many relevant advantages that can be used for selection. In this case, the emitters may not be very broadband, which, when using cascades of emitters, can lead to the saving of individual or many elements. It can be useful for ;" to, for example, reduce the cost of the device. On the other hand, with the same device configuration, higher flexibility and adaptability of the system can be achieved. Thus, it can be, by choice, an increased speed of information transmission by means of the fact that when refusing to use lower frequency channels -And the duration of the cycle can be reduced, while when refusing to use high frequencies, a greater communication range can be achieved . There are a number of good reasons for the need to increase the number of digital states and, along with this, to increase the information density on each information frequency channel.

The pROM method allows for further development in such a way that, in addition to the direction, different slopes of linear phase gradients can also be used, for example, which can be used for coding with the help of frequency shifts of information frequency channels of different magnitudes, and in a specific case, depending on the level of discretization that is achieved, other possibilities of coding and combining may also appear.

In connection with the computer method, positive and negative phase gradients can also be carried out with the help of such changes in the frequency of information frequency channels, which are not exactly proportional to the change

F) frequencies of the main tone (see Fig. 10 below). ka To distinguish it from the pPROM-method, this option is called the pROM-method, and the small n stands for the word pop-rgorogiopa! (disproportionately). The PROM method can be implemented in such a way that the frequency of the corresponding information frequency channel changes in a given cycle somewhat faster or slower than required, for example, by the basic version of the proportional computer method. Obviously, both options

The ROM method is successfully applied on the basis of harmonic frequency series in combination with proportional

by the ESM method.

The principle of an alternative method for the production of phase gradients consists, in other words, in such a modification of the proportional ECM method that an additional small, of course, linear, frequency gradient can be carried out individually for each component during each cycle. Unlike the pRoOM method, when adding with the main tone, instead of a linear phase shift, a slightly different characteristic is formed, namely, of course, a quadratic curve describing an accelerated shift of the phase angle, the direction and shape of which, however, depends on the position of the initial and final values corresponding change in frequency relative to its proper one. With the pROM method, already on the basis of the signs of the first and second derivatives of the relative phase angle, as a function of the cycle time of the main tone, up to six configurations can be distinguished. Taking both options together, a total of up to eight different combinations of signs will be formed for the ROM method.

It is also interesting in the pROM method that, in addition to the signs, certain relative phase angles can also be used, for example, the phases of the initial and final values or the intersection point on the curve of the proper /o phase values according to the KRUUM method.

In Fig. 11 presents a reference version of information encoding (modulation) by one of the transmission devices. The scheme reflecting the basic principle has a device providing information that is coded (modulated) by an encoder (modulator) 3. The encoder converts the information supplied to it into the required code, according to the number of reference and information frequency channels, and delivers coded information 75 to the device ( in the form of generators 5), which generates at least one reference and at least one informational frequency signal, controlled by a special control device. The wave components of the corresponding amplitude, frequency and phase created by these generators are fed to the summing device 9, which is controlled by the control device 7.

According to this embodiment, the signal formed by the summing device is fed, if necessary, to the power amplifier 11, which transmits the information signal to the emitter (or cascade of emitters) suitable for use in the specified environment.

According to this form of implementation, one generator is provided for each frequency channel.

In Fig. 12 shows an embodiment of one of the transmission devices, which performs, if necessary, one of the types of amplitude modulation of the information signal. For this purpose, before submitting the information signal consisting of their separate components (reference and information frequency components) to the summing device, the information signal is first transmitted channel-by-channel to the corresponding modulators controlled by the control device.

In Fig. 13 shows one of the forms of implementation of the receiving device of the claimed system.

A transducer adapted to receive signals in the appropriate transmission medium receives the incoming information signal shown in FIG. 13 as an acoustic signal, and feeds it to the amplifier 23. After the amplifier, a filter 25 is provided, designed for separation and analysis of individual frequency channels and, especially, for passing the reference frequency channel. The signal coming from the filter, preferably a low-pass filter, is transmitted to the frequency detector of the reference channel 27, which estimates the value of the level and frequency of the reference frequency channel. This data is transmitted to the control module 29 of the receiving device. In parallel with Fr

Therefore, before entering the low-pass filter 25, the information signal is also fed to the controlled (accompanying) filters 31, provided one for each of the components. The signals emitted by the filter are analyzed in the controlled boundary device 33 and transmitted to the decoder 35, which will convert the code into an output symbol.

In Fig. 14 presents the following form of execution, which has additional phase detectors, for example, for " VROM, pROM, differential-phase or phase gradient method. For phase analysis, a phase detector 32, located between the controlled filter 31 and the controlled limit device 33, is provided, preferably according to the number of active information frequency channels, and to determine the mentioned ;" phase values, the reference channel processed by the fundamental tone detector is used as a reference.

In terms of signal processing advantages, Figures 15-24 detail various forms of

Implementation of options for signal processing. - And the claimed method contains two essential support functions that can be combined in any way, or used independently. These reference functions are defined as 1) full Doppler shift compensation, abbreviated as MOC, and 2) "channel cleaning", abbreviated as CC. The relevant basic principles are described first -And separately; then various possibilities for combination and improvement of the method and system are explained. 1. Solving the problem of Doppler shifts. o In order to explain the basic principles of MOK, an example is first chosen, in which the frequencies of the reference and i" information channels make up a harmonic series, that is, these frequencies are in integer ratios to each other. If the transmitter varies its frequency, then this happens according to the pUMT method. In the receiver, the separation of all frequency components of the signal occurs immediately at the first step, for example, with the help of Cascade band-pass filters. For simplicity, we first assume ideal transmission conditions such that each component consists of only one single multipath component, and all signal components

F) are emitted and received at approximately the same power level. Such communication conditions can arise, for example, during the propagation of electromagnetic signals in the air environment. This example first explains how and what is achieved with full Doppler shift compensation. 60 The problem of Doppler shifts is that due to the movements of the transmitter and receiver relative to each other, frequency shifts occur that cannot be precisely predicted in advance, since, for example, the speed of relative movements is not precisely known. At the same time, the phase positions cannot be determined with sufficient accuracy, which leads to significant limitations for any form of communication using phase coding. This problem can be partially solved with differential 65 phase coding, in which the actual phase angle is not taken into account, but only its change from clock to clock. One hundred percent Doppler shift compensation can, however, be achieved if the individual information components are processed appropriately together with the reference component. One useful solution includes "pairwise Doppler alignment", abbreviated as pPA, which can be implemented in a variety of ways.

One of the simplest possibilities for this is explained in more detail in the following example:

Let any one, for example, the first information component (the first overtone of the main tone), is presented for processing in one of the time steps selectively and representatively for all others. The frequency of this component Uj, and, therefore, the rate of change of the phase angle b, represent values twice as large as the corresponding values of the fundamental tone uj and 92. Let the signal be received in a discrete form, then the components of the signal radiated by the transmitter zep , and zep;, can be represented in this form: 2Е 2 ИК) zepacip- DE sov pe Kosiv 2E 2 (2) zepak(n) - sove vk veka where M - the total number of digitization of points (counts) on this measure, n - the number of the current count, 5 - the interval time in which digitization took place, pi 5 - discrete time, E - energy, 9,, - initial phase, b - phase angle encoding a certain digital state on the information channel and factor K describes the steepness of the frequency change

Components when applying the proportional UMT method.

In the general case, K can be any suitable function of time, can take positive or negative values, or be equal to zero. The latter means that the use of constant carrier frequencies is considered here as one of the possible special cases.

Since the value of the initial phase of the reference frequency does not change on the transmitting side, it does not play any further role, therefore the corresponding value in equation (1) is set to zero.

As a result of the Doppler shifts, the components of the received signal etri and etri differ from o) radiated only by one additional term: where O is the Doppler coefficient, which describes the ratio of the relative speed of the transmitter-receiver-transmitter (positive sign for mutual convergence and negative for removal) to the speed of signal propagation in the transmission medium.

On the basis of the various arguments of these equations, it becomes clear that the Doppler effect affects these components in such a way that the corresponding members of these arguments differ from each other exactly by a factor of two, which, in fact, is determined by the ratio of proportionality between the frequencies of these components. In this example, the factor of this proportionality is equal to two. no) c Since the proportionality factor is a known value, the exact values of Doppler shifts do not play any more practical role. Precisely, if the reference component is transformed so that it receives the same frequency characteristic as the analyzed information component, then the values of the Doppler shifts of these components will become equal values. In the presented example, with the help of quadrature 49 of the reference component, such reference of Her can be made, which has such a Doppler shift as

It is an informational component. After squaring the reference component, we will get:

Ge) After the unused side band of the spectrum will be filtered, and also - and P - etrid(y)x etribs - ЧЕ sovropiv cats: Oopidih with 50 M x «BE sovsopiv ni Kostiv? vOopiv)- ZE Lsosch(oz sov, and pis kovi: Ogopis chz» M - -2 zhm42 in after scaling the result using the factor -RME, we obtain a normalized signal-link KE whose phase value differs from the phase of the information component presented in equation (5 ) in the following 959 way; o | 2 2 (6)

EGTY - (reovkyut kp - Ogopie) 60 M - p. nie and

This reference signal can be used to some extent as an internal clock of the composite signal, with the help of which the phase angles of the information components can be determined.

Similarly, from the reference component, the necessary reference signals can also be carried out for all other information components contained in the received signal. In this case, the reference component ve must be reduced to the appropriate degree (increased several times by itself) and, if necessary, filtered. In general, the information components can also be transformed in a similar way, which can be useful if the frequencies of the information components are not chosen as in the previous case, but differently, for example, when they lie in a lower frequency range than the reference component, or if their frequencies are not in an integer ratio to the frequency of the reference component. In the latter case, the above-described approach can be applied individually for each side of the pair formed for current processing (the reference and one of the informational components) repeatedly, and exactly as much as is necessary to fully cover both parts of the pair. Since after each multiplication (exponentiation) the number of frequency components of the spectrum increases rapidly, it is necessary to constantly strive for such an arrangement of frequency channels that, if possible, a smaller number of steps will be required for pairwise processing of the received 7/0 signal to equalize the Doppler shifts.

In general, when using applications with phase-coded signals, in the selection of approaches for pairwise equalization of Doppler shifts, it is also necessary to strive to ensure that there is no loss of information during the manipulation of information components, for example, due to the occurrence of uncertainties (ambiguities) in phase position estimates.

Further, on the basis of the mentioned example, one of the simple possibilities is described, how the position of the phase angle of the corresponding information component can be determined according to the above-mentioned approach.

For this purpose, it should be described, for example, the possibility of decomposing the corresponding component on the basis of the quadrature functions of the signal-reference KE |n|.

Due to the fact that the reference CI n) in this example is already presented in cosine form, we write:

KttTeKA p

The corresponding sine component can be obtained, for example, by taking the first derivative of КИп| and corresponding normalization of the amplitude.

Now let's write down the projection of the information component onto the cosine component:

Ia ma ?E 2 sch dv So - U etrik(y)h BOM) - Uk soviyuti cats) Ogo ek adlh

M M (8) 2 x ЧЕ sovyatie zklte Oda zu so 4opiv - Kastiv)e Op ekw ey

И -- -6BM M 2 -ШШШШШШЩШ8ООНШБ5ШЯВ where Mu describes the beginning and Mo the end of the corresponding measure. t 3o Since in the second part of the sum, the value of the function oscillates around zero, then the positive and negative Ge values cancel each other out, that is, this part goes to zero and, therefore, it can be neglected: m

Sa ve LE sova tnep)

The projection of the received information component onto the sine component is written accordingly: o

Che IC 7E 2 - va - Zhetrin(ph RY) - In dreoch oti In coffee(half) I bl epei VIK an x

M IC!

M2 x Uk etsyuti" incl.

M -

More More - -8 UA apseik-war-ya U apsvik- eco a Y SHK!

Next, we consider the values of СО and 50) as, for example, the coordinates of a point in the radial coordinate system. Then, the connecting line between this point, the origin of the coordinates and the abscissa is this desired angle. It can simply be -AND determined using the appropriate algorithm. The obvious method of determining the angle is shown as: d) in -actale, 7 akiap WITH THE QUALITY of euk-vir

Esokek-ep

Ш- The phase of the received component is represented here as the difference between the initial phase of the transmitted wave and some

GI 20 by the encoding phase, i.e. within each clock, it is invariant with respect to time. For the sake of completeness, an option is presented where the difference between the phase in the current and previous clock can be used for

T" coding. Using the indices and and i-th, we will get a variant of differential-phase coding: el (in dec in dr

In a similar way, the phase angles of other information components are determined on each cycle. This gives the user the possibility of correspondingly more accurate (multi-stage) discretization of phase angles and

HF) increasing the speed of information transmission. The above-described method of determining phase angles is defined here as the 7 C-5 projection.

The MOK method contains especially the basis of the variant of the method according to application 28. Fig. 23. demonstrates an overview of the most important elements of processing performed according to the hOK method. The review also demonstrates that different (from the described) elements can similarly find their application for other forms of method implementation. 2. Channel cleaning (CC):

KK contains the identification (in each case) of the most suitable multibeam component and its technical separation from other components while simultaneously minimizing intersymbol interaction. It can already contain a partial compensation of Doppler shifts, which can already be sufficient for a number of different applications.

Consider as an example the case when Doppler shifts do not play an important role; the received signal contains only the interference represented by the multipath components. Communication conditions of this kind are often encountered in acoustic communication between stationary or slowly moving underwater objects. Each of the signal components is then represented by a whole spectrum of multi-beam components. The proposed method should, first of all, provide the possibility of such signal processing, in which the intersymbol interaction is significantly reduced.

Despite the fact that as a result of using the MMT method, different multi-beam components arrive at the receiver with different values of instantaneous frequencies, it is practically hardly possible to separate the most favorable components from the spectrum of the corresponding component immediately at the first step of processing, because they, of course, are very close to each other and their frequencies are time-varying quantities.

Even accompanying filters can hardly be used for this purpose (cannot be configured to work in a narrow enough band). Above, however, we assumed that the corresponding spectra of the reference and information components can be separated from each other (Fig. 176 and Fig. 17c).

After multiplying the reference and the corresponding information component (Fig. 17e), two spectra of intermediate frequencies are formed, which lie in different frequency ranges and have different frequency gradients (Fig. 17e).

The lower frequency range can be passed through a low-pass filter for further processing. In this part of the spectrum, there is also a decrease in the influence of Doppler shifts, while in the other part of the spectrum, these shifts increase their influence. If this second (side) frequency band does not cause 2o interference, it can still remain in the spectrum (unfiltered) at this stage, and one of the filter stages can be spared.

At the next processing step, at least one part of the remaining spectrum is multiplied by the auxiliary accompanying frequency generated in the receiving device (Fig. 171), whose characteristic is chosen so that as a result of the multiplication, one part of the second intermediate frequencies no longer changes in time (fig. 179). high school

The characteristic of the corresponding frequency (Н17Т; Н2; 0... НМ) is either established based on the operational coordination of the signal structures used by the receiver and transmitter or is determined from the previously i) sounding of the properties of the transmission channel (see below "tuning to the channel/").

Fig. 19 demonstrates that this degree can also be achieved when initially only the reference component is separated from the group of information components. With the help of the appropriate selection of the accompanying heterodyne frequency, the frequency provided for the processing of the information component (the first in the current example) can be stabilized. with

The advantage of this approach is that, with the help of a suitable accompanying heterodyne frequency, M, the necessary parts of time-invariant (stable) frequencies can be placed in defined frequency windows and, at the same time, filtered using conventional (non-accompanying) filters, such as low-pass filters, me) z5 (Fig. 17p). Cha

Fig. 20 shows on an example (close to practical use) that due to the existence in the spectrum of a whole series of stable intermediate frequencies of multi-beam components of the signal, no conclusions can yet be made about the value of the phase angle, because different multi-beam components, represented at different moments of time by different power levels. "

For this reason, a second stage (with the best possible quality) of filtering is provided, the bandwidth of which is set (for each component) on the basis of the data on the channel properties obtained during the "channel tuning" procedure (see the description below) on the most powerful at the level of the multibeam component. ;" The dashed line (Fig. 172) shows that the slope lines of the front and back edges of the filter can be set with the required steepness. Therefore, the influence of other multibeam components can be significant

Decrease (fig. 171). -I Fig. 21 presents a practical example of how, as a result of such narrow-band filtering, the influence of fluctuating multi-beam components (see Fig. 20) can be significantly weakened (by suppression of these components) and, as a result, an unambiguous assessment of the information parameter can be made. In general, the process described in this context is defined as "channel cleaning" with partial compensation of Doppler shifts.

Determination of parameters: The signal components prepared in this way and cleaned from the effects of channel interference can now be subjected to a detailed analysis of their parameters. At the same time, both the amplitudes and phases of the information-carrying signal components can be determined with high accuracy. Different values of amplitudes can be distinguished, in the simplest case, by means of boundary devices. In the event that C-5 projection is necessary to determine the phase angles, the components of oscillation-reference (sine and cosine of oscillation) necessary for decomposition can be (artificially) generated by the receiving device. The latter does not represent technical

F) complications, since the settings of the last stage (narrowband) filtering are known to the system and, therefore, the frequencies of the information components of the signal are known. Depending on the used modulation (encoding), the user can choose and apply the most suitable forms of processing from the repertoire of well-known algorithms.

In the above-described method, the procedure of "cleaning the CC channel" is mainly used together with the pRUMT method (compare with figure 5). This method can also be implemented without problems together with the rauMtT method (compare with fig. 15). In the case of using the ramMT method the product of the reference and information components leads directly to the formation of stable intermediate frequencies, while the further 65 Multiplication with the auxiliary component becomes redundant. by multiplying by an appropriate auxiliary component of constant frequency This is, however, within the scope of the above description.

CC channel cleaning is compatible with all forms of the MMtT method, in which the gradient of the frequency change is greater than zero. In order to distinguish the described option from the following modifications, it will be called the KK1 method. A useful embodiment of the claimed method described here forms the basis for clause 18. The most important elements of this method are presented once again by looking at FIG. 23.

The above-described variant of cleaning the CC channel can be transformed, for example, so as not to first multiply the information component by the reference component. In this case, the formation of 70 stable intermediate frequencies occurs by multiplying the corresponding component by the corresponding accompanying auxiliary frequency. This approach gives such an advantage that the spectra of stable intermediate frequencies do not have such complex spectra of multi-beam components as the adopted components. After separating (filtering) the best multi-beam component of each of the components (channel cleaning), there is still an opportunity to process the information component with the help of a link in such a way as to achieve at least partial compensation of 7/5 D-Doppler shifts, or how, by analogy with the example described for mOK, to carry out estimation of the phase angle of the information component using C-5 projection to form the sine and cosine of the component from the "purified" reference component. The selection of the frequency response of the corresponding link takes place, if necessary, during multiplication with the corresponding auxiliary accompanying frequency, or after passing the last stage of filtering by means of multiplication with the corresponding auxiliary constant frequency. In the latter case, only one pass of the reference component through the filter is necessary.

The following paragraph describes modifications that determine further useful forms of implementation of the invention.

Schematically simplified processing plan is presented in overview figure 23 and named as KK2-method.

If Doppler effects do not play any role, the reference component can be omitted or used as an additional information component. In this case, the QC2 method is used. high school

The determination of the signal parameters should, however, occur as described for the CC1 method.

For the sake of completeness, let us once again refer to the above-described, but not figuratively presented, alternative solution, by means of which i) the stage of obtaining stable intermediate frequencies can be achieved within the framework of, for example, the ruUMt method without preliminary separation of components simply by multiplying signals received directly in adjacent measures. This step also involves partial compensation of Doppler shifts. What is special in this case is that, depending on the steepness of the frequency change, the spectra of stable intermediate frequencies of the corresponding channels lie in different, although more or less close to each other, adjacent frequency windows. However, as a result of such a transformation, a very complex signal structure is obtained. Especially, M if a large number of information channels are used, it is necessary to carefully monitor the possible overlap of the products of multiplication of the processed components. A cascade of narrow-band filters can be used, for example, to separate multi-beam components. Cha

In conclusion, it is necessary to point out once again that only the principle of the proposed approach is presented here. In practice, it is possible to use more complex methods of signal processing and analysis, which include the use of the above-described work steps of a similar or different form. The principle remains, however, the same.

Complete solutions: "

After the methodical bases for full compensation of Doppler shifts and different options for cleaning the canal (which also include partial compensation of Doppler shifts) have been described separately, we will consider

And a practical case in which interference during signal reception causes a lot of multi-beam interference? components, as well as Doppler shifts. This combination of interfering factors often makes communication between objects moving underwater difficult.

For this example, there is a possibility of obtaining a solution, for example, by combining the mOk- and -I KK2-methods:

After the separation of the reference and information components, first perform, as described for the MOK method, a pairwise equalization of the Doppler shifts, in which at least one of the components of the selected -I pair will be transformed (or, if necessary, both components will be transformed) so that both components of the pair will accurately represent the same characteristics of the frequency change, and therefore are equally "loaded" with Doppler shifts. If necessary, the frequency bands (components) of the signal that are not necessary for processing can be filtered out and the remaining signal components can be normalized again.

After that, both components are separately multiplied, usually with the same auxiliary accompanying frequency, (which has the same frequency change gradient as the corresponding component, but is parallel shifted by some amount), while stable intermediate frequencies are formed, which are then filtered to make channel cleaning in subsequent filtering steps. For this, narrow-band filters are installed for each

F) component separately. In an ideal case, the appropriate filter settings can also be taken into account in the exact matching of the aforementioned auxiliary accompanying frequencies.

As a result, the cleaning of intersymbolic interactions is achieved, both in the reference component and in the information component. After such cleaning (including, if necessary, the analysis of limit values), the analysis of the parameters can take place, for example, in accordance with the described approach for mOK or KK2, and with the help of pairwise processing of the corresponding information component with the corresponding reference signal, full compensation of Doppler shifts is achieved.

The above-described embodiment of the claimed method forms the basis for a useful embodiment 65 of the Method according to claim 8. In a schematic view, in Figure 23 it is designated as Cotri. 1.

The next possible solution consists in the appropriate combination of the хОкК and КК1 approaches (see the simplified representation of the processing scheme in Fig. 23):

Also in this case, after separation of the reference component from the information component, the Doppler shifts are equalized in pairs. After that, one of the components is shifted parallel to some

Frequency value by multiplying by the corresponding auxiliary constant frequency generated by the receiving device. Both components are then multiplied, thereby achieving the processing step shown in Fig. 179, that is, the stage of formation of stable intermediate frequencies. After that, filtering (at both steps) and analysis of parameters according to the QC1 method is carried out.

This second (full) solution assumes that by projecting the information component onto the 7/0 signal-link, which has an identical characteristic of Doppler shifts, the effect of frequency shifts caused by the movement of objects is completely eliminated. At the same time, this reference signal is "used up". However, it will no longer be needed. A significant advantage of this approach is that in order to place the necessary part of the intermediate frequency spectrum exactly within the given frequency window, there is a need only to generate an auxiliary constant frequency. Under favorable conditions, the same auxiliary /5 frequency can be used for all pairs of processed components. In principle, there is a possibility of using this auxiliary frequency at the final stage of cleaning the channel as a reference signal for analyzing phase values. Since in practice, they try to set narrowband filters for each signal component individually (in this case, a combination of the reference and the corresponding information component is already presented), then the current settings of the filters become known to the system in each case of debugging, and therefore the receiving device can artificially generate agreed with such corrections of the reference signals (including quadrature sine and cosine components), if they are necessary for the analysis of phase angles.

Channel installation and channel debugging:

With the help of the repeatedly mentioned tuning per channel, on the basis of the corresponding test signals c, it is first necessary to establish that the signal structure is optimally selected for the corresponding communication conditions and/or it is ensured, at least, that the receiving device can separate the signal components i) to the required extent . If this prerequisite is met, then channel debugging can be performed for all embodiments of the method conforming to point 1 and also for all subsequent variants requiring channel cleaning.

For this purpose, the transmission of initially longer non-coded signals can be envisaged, which otherwise, however, have the necessary characteristics, which are used further for the transmission of information. At the same time, the user can decide whether all frequency channels are used at the same time, or whether tuning on channel c is carried out on the basis of test signals that are sent in turn and represent in each case either only the reference component or only one of the information components. Any of these approaches must, of course, be compatible with the chosen form of implementation of the signal processing method. The received test signals about c5 go through all the processing steps provided by the corresponding method in order to generate stable intermediate frequencies. Cha

At this level, an analysis of the energy distribution in a given frequency spectrum is carried out individually for each of the signal components (or from the mixed components obtained by multiplying the reference and the corresponding information component) intended for further processing. For such an analysis, fast Fourier transformation can be used, for example. Based on the results of the processing, the best (favorable) multi-beam component (which, of course, has the highest power level) is selected, for which a qualitative setting of narrow-band filters is then performed in c, the results of which are stored in memory. After completing the appropriate settings for all signal components, the actual data transfer can begin;" (information). These settings are saved until the next channel debugging.

Especially during acoustic data transmission in underwater channels, communication conditions are often shaky and changeable

In time. In such cases, it is recommended to carry out the tuning procedure on the channel again after -And time intervals corresponding to the communication conditions, which means that the narrowband filter settings are regularly updated. o Using longer uncoded debug signals per channel offers good static-I reliability, although it means that the actual data transmission must be interrupted from time to time. However, such interruptions of 5 years can be avoided. A useful alternative is the method stated in point 23. At the same time, on the basis of the received signals, tuning to the channel with the restoration of the mentioned filter settings is carried out in parallel with the actual processing of the signal (or as a part of it).

At the same time, it is advisable to connect to the processing of the results of receiving signals from a large number of clocks. One of these alternative solutions sets correspondingly more complex requirements for the signal processing system. Determination of the relative speed between the receiver and the transmitter: This part briefly presents how, with the help of the received signals, an assumption can be made regarding the current change in the distance between the receiver and (F, transmitter). , phase angles) for each of their signal components. The signal components are processed together in such a way that Doppler shifts are eliminated. The latter are considered as interference quantities, because Doppler shifts contain, however, information in the form of the Doppler coefficient O - M /s, which although has nothing to do with the actual data transmission, but can, however, provide information about the value of the instantaneous relative velocity M between the receiver and the transmitter. The Doppler coefficient can be determined using a suitable signal processing method. Since the signal propagation speed c is a known quantity (for example, it can to be measured in the process of nding of the channel), the value of M can be estimated and relatively accurately 65 Determined.

As an example, the following possible solution can be presented: For this purpose, any one component of the received signal can be filtered (it is advisable to select the non-encoded reference component). If necessary, again, according to the CC2 method, it is possible to reduce the spectrum of the received signal to one multi-beam component. Since the signal structures are known to both the receiving and the transmitting device, and it is also possible to determine the phase angle in the process of analyzing signals using one of the familiar methods, the system can generate a reference signal normalized by amplitude, which in terms of its phase angle and frequency response - with the exception of the not yet known Doppler component - is identical to the corresponding component of the received signal. After taking the projection of this component of the received signal on the quadrature sine and cosine that make up the reference signals and after filtering (low-pass filtering), the pure 7/0 Doppler component can be obtained in the form of sinusoidal and cosine oscillations of equal amplitude. The arctangent function supplies the argument ЮООпи 5. Since Op; is a known value, the distribution. of these quantities gives 0, and ЮО multiplied by s gives, in the end, M. (Schematic plan see figure 23 under Oorrieg-Vevi.).

For many applications, it may be useful to obtain this useful information without additional measurement costs.

Next, it should be noted that the knowledge of the Doppler component can contribute to the further improvement of the signal processing itself. Thus, for example, the auxiliary frequencies generated within the CC can be more precisely matched with the structure of the corresponding component of the received signal and, thus, unwanted (in the actual analysis of the signal), Doppler shifts can be attenuated better or in a simpler way. With the help of the integration of such measures, if necessary, their iterative application, in addition to improving the processing results, at least an intermediate methodical optimization can also be achieved, so with the increase in resistance to Doppler shifts, the areas of application can be expanded, for example, KK1 and KK2 methods. At the same time, it is possible to avoid, especially when multiplying the components of the received signal with each other, the rapid growth of computing costs, in some cases it is possible to abandon the intermediate step of filtering and perform the processing in general faster. All the simplifications of the central part of the method are useful for achieving real-time processing modes. Even if the estimation of Doppler shifts initially causes additional computational costs, overall savings can still be achieved, because then the central procedures of i) signal processing can be performed with the use of less software and/or hardware work.

Further, the mentioned simplifications or improvements of the claimed method can be achieved in a simple way also in the case when the information about the current Doppler shifts is provided in one of the forms, for example, from "g" on the part of the external measuring system.

In Fig. 24 shows the functional diagram of the signal processing device. It includes a filtering device, consisting of two parallel controlled filters BPE1 and BPE2, which divides the received signal into M reference and information components.

Both parts of the signal are first fed to the frequency conversion device (consisting of UUapaieg!i and me)

Uapaiegg2), in which the pairwise equalization of the Doppler shifts takes place, before these two parts of the y-signal are multiplied by a multiplier of 1, in order to then be converted by an auxiliary (heterodyne) carrier frequency (processing system supplied by the generator) and a factor of 2 into constant (no longer changing over time) intermediate frequencies.

As a means of suppressing component interference (in this form of implementation of the method), first filter I PE1 and, if necessary, also a second filter ЭР2, which in each case are included after the corresponding multipliers, to remove side frequency fragments unnecessary for processing from the spectrum. Then the multi-beam component suitable for processing is separated from the spectrum with the help of the included in the chain;" VREZ filter processing; the signal is then fed to the parameter analysis device (in this form of execution of the parameter analysis module). If necessary, the parameter analysis module can be supplemented with a reference signal generator connected to the VREZ. -And at the end of the processing device for each information component, the signal parameters used in the modulation of the transmitted symbols (for information encoding) are issued. about Fig. 25 represents a functional scheme for debugging per channel, which can be advantageously used in a similar context. In contrast to figure 24, in this embodiment, the signal components after the ГРЕ2 5р filter are fed to the tuning device (tuner), in which the PPT module (or module for frequency spectrum analysis) and the device marked with the number 3 represent a processing device. The result is fed to the control device, which performs the appropriate optimal adjustment of the VREZ filter.

Next, further possibilities of application of the claimed method and system are described in detail.

When reducing the distance between the receiver and the transmitter, additional frequencies can be optionally used

Channels lying between the frequency channels used initially, and higher frequency channels can be used, or the entire frequency spectrum can be shifted entirely in the direction of higher frequencies.

Ф) At the same time, such an additional beneficial effect appears that as the communication distance decreases, the influence of interference factors also decreases. To implement this, the receiver and transmitter must have the ability to utilize a fairly wide range of frequencies, as well as the ability to adjust the modulators bo (coding devices) accordingly. On the part of the transmitter, the recognition of the additionally entered frequencies can be carried out either automatically, or the transition to a new operating mode can be made using the transmission of a message from the transmitter (for example, with the last information package). The individual tone channels must, however, be so far apart that they can be separated by the receiver under appropriate communication conditions. On the other hand, increasing the communication distance may cause the need to shift the spectrum of the used frequencies to a more low-frequency range or the need to increase the frequency distances between channels (especially when the influence of interference increases) by means of proportional stretching of the signal spectrum or by means of dropping intermediate frequency components.

The choice of frequency ranges, provided to a greater extent from the basic options, so that the tones or frequencies of the channels have a consonant or integer ratio (overtone) to the fundamental tone, which has the lowest frequency, is aimed at achieving an energetically favorable ratio.

When applying harmonic frequency series, there is, in addition to everything, the possibility of using non-linear effects during the propagation of sound vibrations and at the same time achieving long information transmission distances.

Sound waves are longitudinal waves in which areas of high density alternate with areas of low density.

Since the speed of sound depends, among other things, on the density of the medium, denser areas propagate 7/0 faster. The fronts of the initial sinusoidal oscillations become gradually asymmetrical, that is, the sinusoidal oscillation will turn more and more into a sawtooth oscillation. Physically, this means transferring energy to overtones. In water, this effect is felt at a distance of several kilometers. If overtones (one or more) are sent simultaneously with the main tone, then the latter, due to harmonic relationships and the mentioned nonlinear effects, receive additional energy from lower-frequency tones. As a result, they disappear in the noise 7/5 not so quickly (stay longer above the background noise level) and therefore reach long distances. Since the data transmission distance of the system is primarily determined by the highest frequency of the system, the overall communication radius increases. It is advisable to emit the main tone (if possible, also other tones) with high power.

Due to the high variability of the system, other properties of the transmission channel may also be used. As a result of the layered heterogeneity of the water medium, 2o transmission channels are often created, which have their own characteristics of the propagation of oscillations. Depending on the corresponding eigenvalues, slightly different modes can be excited, which, as a rule, have relatively low frequencies, but do not propagate very long distances. In the information transmission system, in principle, there is a possibility of matching the frequency ranges to such modes. For this, the receiving and transmitting devices must be able to agree on such an exchange mode. with

In the event that the relative speed of the receiver with respect to the transmitter is so small that Doppler shifts can be neglected, appropriate uniform frequency changes may well be performed for the entire system as an alternative to the proportional BOM method described above. In this case, a so-called curve of frequency shifts, or, figuratively speaking, a "melody" can be set (depending on the specified communication conditions) and composed with all frequency channels. This method is called the equivalent ECM method. The peculiarity of this method is that as a result of parallel shift of frequency channels, a constant frequency gradient will be formed, i.e. frequency shift speed, and in the optimal case, the optimal frequency resolution of multi-beam components (interference) can be achieved for the corresponding channels across the entire frequency spectrum. M

The method modified in this way has, among other things, the advantage that the spectrum of frequencies occupied by the signal components does not expand with an increase in the reference frequency. Based on the greater compactness of the location of the

35 Component in the spectrum, the upper tones are unlikely to be in danger of appearing in a frequency range with too small a radius of oscillation and therefore being partially cut off with loss of information. Upper frequency bands (more favorable in terms of data transfer rate) can be better used. Collateral

The ESOM method can be implemented more easily, because emitters and receiver sensors can often work only in a very limited frequency band, and the use of appropriate cascades is not always possible. "

It goes without saying that even with the parallel ECM method, the receiving device must be informed how individual frequency channels should be matched in relation to the main tone. In principle, switching between the equivalent and proportional ECM methods should not be problematic, since in the sense; determining the frequency characteristics relative to the reference channel, it is only necessary to move from the multiplication function to the addition function.

If phase jumps are causing problems at the clock coupling points, it may be useful to implement -I clock-by-clock amplitude modulation. One of the further ways to minimize the effects of interference consists in applying such a coding approach where the existence of tones in two adjacent beats of the same information channel is excluded. The same effect can be achieved by means of multiplexing in -And alternating use of even and odd information channels. The fact that, in addition to other parameters, the speed characteristic (depending on the frequency value) can also be determined with the help of special sounding or in the process of two-way communication, and the fact that it can (and even should) be taken into account when generating signals is itself which is understood

If necessary and if the current conditions of communication allow it, the use of the reference frequency channel is additionally foreseen as an information frequency channel.

F!

Claims (1)

The formula of the invention is) 1. The method of information transmission, in which the transmitted signal includes at least one reference component (VC) and at least one information component (11, 12, ..., IM), and - at least one of these components is carried out continuous frequency change during the information transmission session and, in addition, the reference component (VC) and the information component (11, I2, ..., IM) form appropriate discrete states for the formation of a bit pattern. 65 2. The method according to point 1, in which at least one reference component and at least one information component continuously change the frequency over time, and a certain ratio is set between the frequencies of the reference and information components according to a predetermined function of time. C. The method of claim 2, wherein the frequency spacings between the components are constant or proportionally varying. 4. The method according to any one of claims 1-3, in which the frequency of at least one component during the data transmission interval continuously increases. 5. The method according to any one of claims 1-4, in which the frequency of at least one component continuously decreases during the data transmission interval. 6. The method according to any of the previous items, in which the gradients of frequency changes are set depending 70 on the position of the interference frequencies with respect to the corresponding frequency of the signal component to minimize intersymbol interference and/or appropriate offsets are set to prevent harmful mutual interference of several data transmission systems. 7. The method according to any one of claims 1-6, in which the initial frequencies of the components change their values from one data transmission interval to another. 8. The method according to any one of claims 1-7, in which the frequency change ranges, i.e. frequency bands, of two or more components overlap. 9. The method according to any of claims 1-8, in which at least one component, preferably a reference frequency, lies in a separate frequency band. 10. The method according to any one of claims 1-9, in which the bit pattern is set preferably during a given 2o time clock by changing the frequency, amplitude and/or phase or dynamic phase characteristic. 11. The method according to any one of claims 1-10, in which the bit pattern is changed during one time clock. 12. The method according to any of claims 1-11, in which the number of information components (I71, 12, ..., IM) varies depending on the data transmission channel. 13. The method according to any of claims 1-12, in which the reference component (VC) is used as an additional information component (IM-1). 14. The method according to any of claims 1-13, in which the reference component (VC) and at least one information and) component (11, 12, ..., IM) are formed in the form of a sound or electromagnetic wave. 15. The method according to any one of claims 1-14, in which to process the information signal after reception, the reference component (VC) is separated from at least one information component (11, 12, ..., IM). 16. The method according to any of claims 1-15, in which pair processing of the reference component (VC) and at least one information component (11, I2, ..., IM) is performed. with 17. The method according to any of claims 1-16, in which the information component and the reference component or the processed i-pair reference components and the information components are converted to constant intermediate frequencies (71, 72, 2, ХХ) preferably by multiplying with auxiliary frequencies (NI, H2, ..., NMZHKH). and) 18. The method according to any of claims 1-17, in which, preferably in the variant with a proportional change in the frequencies of the signal components, constant intermediate frequencies are achieved by means of pairwise processing, preferably by multiplying the signal received in the current clock with the signal received in to the previous measure. 19. The method according to item 16 or 18, in which to separate the frequency components of the spectrum of the received signal, the effect is used that, depending on the magnitude of the frequency gradient applied in the emitted signal, the difference in the propagation time of the multi-beam components in the channel, after conversion into c of the received signal to intermediate constant frequencies, presented in the form of larger or smaller frequency differences, and the most favorable frequency components for further processing are selected from the spectrum;" of intermediate frequencies (21, 72, ..., 7-X) mainly with the help of filtering means and/or on the basis of the analysis of such components, the corresponding informational parameters of the signal are found. 20. The method according to any one of claims 15-19, in which tuning to the -I channel is carried out at given time intervals. 21. The method according to any of claims 15-20, in which in the process of data transmission, continuous identification of the most favorable for subsequent processing of the multi-beam component of the spectrum of the received signal -I and/or updating of filter settings based on the appropriate analysis of the spectra of intermediate constant frequencies, according to by means of which tuning to the channel is implemented as a procedure that is performed continuously and without stopping the actual data transfer process. 22. The method according to any of claims 15-21, in which the Doppler frequency shifts that occur in the process of data transmission are mainly found by the receiving system itself and are taken into account when generating auxiliary frequencies. 23. The method according to any one of claims 15-22, in which pair processing is carried out using components generated by the system itself, which in each case have a corresponding frequency characteristic. F) 24. The method according to any of claims 15-23, in which in each case of a) the reference component (VC) is transformed into a transformed reference component (VC) and at least one information component (11, 12, ... . (sine and cosine of the component), obtained from the corresponding (VC). 25. The method according to item 24, in which the reference component is transformed by a suitable procedure into a reference component having identical Doppler shifts with the information component, which is processed in each case in such a way that the multiplication of these two components forms a spectral component 65 (signal) constant frequency. 26. The method according to item 24 or 25, in which a) multiplication of the transformed information component (11, 12", ..., IM) is carried out with the reference component (KE) to form some first value (CO), b) multiplication of the transformed of the information component (11 12, ..., IM) with the first time derivative of the reference component (KE) to form some second quantity (50), c) the ratio between the first and second quantities is determined to obtain a time-invariant quantity that depends only on the information parameter, also invariant in time. 27. A system for transmitting information for implementing the method according to any of claims 1-26, containing transmitting and receiving devices, between which an information signal (IZ) is transmitted, and 70 - the transmitting device contains a means for creating a reference component (VC) and at least one information component (11, I2, ..., IM) for generating time-continuous frequency changes and formation of bit patterns, as well as - the receiving device contains means for receiving a signal consisting of a reference component (VC) and at least one information component (11, I2, ..., IM), in which at least one component has a continuous change in frequency over time. 28. The system according to item 27, in which the transmitting device contains - at least one generator for the formation of the reference component (RC) and at least one information component, - the first control module, which is connected to the generator and sets the characteristic of the frequency change over time, - coding a device or modulator connected to the control module to convert information into a suitable signal, as well as a mixer standing in the connection diagram after the generator and the coding device or modulator. 29. The system according to item 27 or 28, in which the receiving device contains at least one input, a processing device and at least one output, and the processing device contains a serial connection of the following means: c - means for separating and converting signal components, especially for their conversion in o constant intermediate frequencies, - means for separating or muting signal components that are interference, as well as - means for parameter analysis. 30. The system according to any of claims 27-29, in which the means for separating and converting the components of the signal "g z o ) with a reference component (VC), and the products of multiplication c form spectra of constant intermediate frequencies, from which the desired components of the signal are filtered by means of a series-connected means for suppressing components that are interference, containing at least one filtering device, which then are transmitted to the serially connected means of analysis of parameters. uh- 31. The system according to any one of claims 27-30, in which the means for separating and converting the signal components, in addition, includes a filter device with a control module, which is in the connection circuit in front of the multiplier, as well as at least two filter elements in serial connection, with the help of which at least one component is separated from other components of the signal. " 32. The system according to item 30 or 31, in which the means for separating and converting the signal components, except from the multiplier intended for pairwise processing of VC and IM, contains an additional device with a module that . produces auxiliary frequencies and, if necessary, with an additional multiplier, these devices provide and? conversion of the signal components - if necessary, bypassing the intermediate stages at which these components still have time-varying frequencies - into predetermined ranges of constant intermediate frequencies. 33. The system according to item 29 or 31, in which the means for separating and converting the signal components contains -And at least one multiplier and at least one module for forming auxiliary frequencies in the form of one or more generators or a polled memory device, with the help of which the reference and o informational components are converted separately from each other into intermediate frequencies that are in the specified -I frequency range, a means for muting the components, which are 5o interferences, is connected in series to these hardware means, which contains at least one filtering means, with the help of which the desired o components of the signal from the corresponding spectra of constant intermediate frequencies, thus the signal itself is freed from interfering components, and then transmitted to the serially connected means for parameter analysis. 34. The system according to any one of claims 27-33, in which the means for separating and converting the signal components additionally includes at least one means for equalizing Doppler shifts. 35. The system according to any one of claims 27-34, in which the means for suppressing components that are interferences includes (F, an additional controllable filter for suppressing components that are interferences. ka 36. The system according to any one of claims 27 -35, in which the parameter analysis means contains at least one multiplier for pairwise processing in each case of one information component of the signal with at least one reference oscillation, which is created in the system by the corresponding generator or taken from the storage device, or provided using the reference component and also contains a module that evaluates the values of information parameters. 37. The system according to any of claims 27-36, which additionally includes a tuning means, which is located in the connection diagram after the means for frequency conversion and, preferably, before the means for analyzing parameters, and b5 also contains a module for analyzing frequency spectra and processing a device combined with a means to suppress components of the signal that are interference. 38. The system according to any one of claims 27-37, which additionally includes a module for equalizing Doppler shifts, which is connected to at least one generator of auxiliary frequencies and/or an additional module for obtaining the rate of change of the distance between the receiving and transmitting devices. se ch o "I se cha Ge) - - with . a -I (95) -I with 50 GT» ko bo b5