RU2723300C1 - Method of signal separation with frequency shift modulation and compensation of combination components - Google Patents

Abstract

FIELD: radio engineering and communication.SUBSTANCE: invention relates to radio engineering and can be used in communication systems. Prior to setting values of frequency shifts between adjacent signals so that value of difference of any pair of frequencies does not exceed value of frequency, for which the difference between the amplitude-frequency characteristic (AFC) of the low-pass filter (LPF) and the band-pass filter becomes less than a certain predetermined value; after multiplication by corresponding reference signals in multipliers, results of signal and interference conversion are formed, each of which is branched into two identical components: a first component, LPF is filtered, whose band is matched with a signal band; simultaneously second component is filtered by band-pass filter, which pass band is selected so that upper frequency of band-pass filter corresponds to upper frequency of signal, lower frequency of band-pass filter is set as close as possible to zero value, signals passed LPF and band-pass filter are subtracted from each other in each line, obtained signals are converted to digital form in corresponding ADC; obtained values are summed up and stored; sum with the maximum value is found from the obtained sums; determining the threshold value by multiplying the found maximum value by a coefficient, value of which is set in advance; obtained sum values are compared with a threshold; based on the comparison results, the presence or absence of a signal with the corresponding frequency is stated.EFFECT: technical result consists in improvement of noise immunity of communication equipment.1 cl, 2 dwg

Description

The proposed method relates to radio engineering and may find application in communications.

Known methods that are implemented by devices for suppressing broadband interference described in patents RU 2115234, H04B 1/10, RU 2143783, H04B 1/10, RU 2190297 H04B 1/10, the disadvantage of which is the low degree of interference suppression.

A known method of separation of signals in the presence of interference, described in patent RU No. 2675386 H04B 1/10, the disadvantage of which is its low efficiency when using a multi-frequency signal.

Known amplitude and angular modulations described in the training manual "Fundamentals of the theory of radio systems. Tutorial. // V.I. Borisov, V.M. Zinchuk, A.E. Limarev, N.P. Mukhin. Ed. V.I. Borisova. Voronezh Scientific Research Institute of Communications, 2004. ”, pp. 165 - 168, 170 - 174, the disadvantage of which is low efficiency under the influence of interference.

Known methods for digital signal processing with pulse amplitude modulation (ASK), quadrature amplitude modulation (QAM), phase shift modulation (PSK), described in the book "Prokis John," Digital Communication ". Per. from English / Ed. D. D. Klovsky. - M .: Radio and communication. 2000, p .: 148, 149; 150, 151; 151, 152, respectively, the disadvantage of which is low efficiency under the influence of interference.

The closest analogue in technical essence to the proposed one is the method, which consists in the use of orthogonal frequency shift modulation (FSK) (multiplexing (multiplexing) with orthogonal frequency division of channels (OFDM)) and signal extraction using the optimal maximum likelihood of the detector described in the book " Prokis John, Digital Communications. Per. from English / Ed. D. D. Klovsky. - M .: Radio and communication. 2000, p. 141, 208, 219-221, 593-596, adopted as a prototype.

The prototype method is as follows.

When using a frequency-shift modulation method, M orthogonal signals of equal energy are generated. These signals vary in frequency.

cos(2πf_ct+2πmΔft).

(1)S _m (t) =

cos (2πf _c t + 2πmΔft).

(1)

Here: m = 1,2, ..., M;

0≤t ≤T;

– энергия сигнала;

- signal energy;

T is the signal change period corresponding to the minimum value of the frequency of the signal spectrum;

f _c is the signal frequency;

Δf is the frequency shift between the signals.

The equivalent low-frequency signal is determined as

e^j2πmΔft, m=1,2,…,M, 0≤t ≤T. (2)S _tm (t) =

e ^j2πmΔft , m = 1,2, ..., M, 0≤t ≤T. (2)

These waveforms are characterized by equal energy and cross-correlation coefficient, the real part of which is equal to

ρ _r = Re (ρ _km ) = (sin (2πt (mk) Δf)) / (2πt (mk) Δf)), (3)

Re (ρ _km ) = 0 when Δf = 1 / (2T) and m ≠ k.

Since the case │m-k│ = 1 corresponds to adjacent frequency intervals, Δf = 1 / (2T) represents the minimum value of the frequency separation between adjacent signals for the orthogonality of M signals.

An additive mixture of signal and interference is received at the input of the receiver

U = U _s + U _p (4)

where: U _s is a signal generated using frequency shift modulation;

U _p - interference.

After multiplying by the corresponding reference signals S _op.i in the units of multiplication and integration by integrators, the result of signal and noise conversion is formed at the outputs of the integrators, i.e. reference signal multiplication and integration (correlation metrics):

K _is U _s , + K _ip U _p , (5)

where K _is , K _ip are the signal conversion and interference coefficients, respectively, depending on the type of system of orthogonal functions used.

In the selection device, the maximum signal is selected that corresponds to the largest correlation metric.

The disadvantage of the prototype method is not high enough efficiency under the influence of interference, which is explained by a wide spectrum of the signal and a high level of conversion noise.

The objective of the proposed method is to increase the efficiency of signal extraction under the influence of interference due to the fact that the use of a signal with a smaller value of the width of the spectrum and a decrease in the noise level of the conversion are ensured.

To eliminate this drawback in the method of isolating a signal using frequency shift modulation and compensation of combination components, which consists in generating a signal consisting of several harmonic signals using frequency shift modulation (FSK), synchronizing means is performed at the stage of entering into communication connection, after multiplying by the corresponding reference signals S _op.i in the multiplication blocks, the result of signal and noise conversion is formed, i.e. multiplication by reference signals, according to the invention, sets the frequency shifts between adjacent signals in advance so that the difference value of any pair of frequencies does not exceed the frequency value for which the difference in amplitude-frequency characteristic (AFC) of the low-pass filter (LPF) and the band-pass filter becomes smaller some predetermined value; Signal processing is carried out simultaneously and in the same way in the respective lines, namely: after multiplying by the corresponding reference signals, the results of signal conversion and interference are generated in the multiplication units, each of which is branched into two identical components: the first component is filtered by the low-pass filter, the band of which is matched with the signal band ; at the same time, the second component is filtered by a band-pass filter, the passband of which is selected as follows: the upper frequency of the band-pass filter corresponds to the upper frequency of the signal; the lower frequency of the band-pass filter is set as close to zero as possible; the low-pass filter and the band-pass filter are selected with phase-frequency characteristics identical to the maximum and so that the frequency response of the band-pass filter in the region of frequencies close to zero has the greatest possible slope; in the frequency domain, starting with a value for which the difference between the frequency response of the low-pass filter and the band-pass filter becomes less than some predetermined value, ensure that their frequency response is identical to the maximum extent; the signals that have passed the low-pass filter and the band-pass filter are subtracted from one another in each line, the received signals are converted into digital form in the corresponding ADCs; the obtained values are summarized and stored; from the received amounts, the sum with the maximum value is found; determining a threshold value by multiplying the found maximum value by a coefficient whose value is set in advance; the obtained values of the sums are compared with a threshold; according to the results of the comparison, a conclusion is made about the presence or absence of a signal with an appropriate frequency.

The proposed method is as follows.

The signals are formed as the sum of n harmonic signals (subcarriers) with different frequencies using frequency shift modulation. The values of neighboring frequencies differ by a certain value Δf _ij .

Here i, j are the numbers of neighboring frequencies, j = i + 1.

The frequency shifts are set so that the difference value of any pair of frequencies does not exceed the frequency value for which the difference in the amplitude-frequency characteristic (AFC) of the low-pass filter (LPF) and the band-pass filter becomes less than some predetermined value (F _p ) (see Fig. .1).

The reference frequencies are formed with the same values as the harmonic signals.

The number of harmonic components n used in the formation of the signal, the values of the frequency shifts between the signals are determined at the development stage by experiment or by mathematical modeling as values that provide the maximum degree of noise immunity at a given level of data exchange rate.

At the stage of entering into communication, the signal is synchronized by any known method, for example, by processing the code sequence used.

After multiplying the additive mixture of the signal and the interference by the corresponding reference signals, the result of the signal and interference conversion is formed in the multiplication blocks. The signal at the output of each multiplication block is branched into two identical components. The first component is filtered by the low-pass filter, the band of which is consistent with the signal band. At the same time, the second component is filtered by a band-pass filter, the passband of which is selected so that the upper frequency of the band-pass filter corresponds to the upper frequency of the signal, the lower frequency of the band-pass filter is set as close to zero as possible.

The choice of the low-pass filter and the band-pass filter is carried out with phase-frequency characteristics identical to the maximum and so that the amplitude-frequency characteristic (AFC) of the band-pass filter in the frequency region close to zero has the maximum possible slope, in the frequency region, starting from a value for which the difference in values The frequency response of the low-pass filter and the band-pass filter becomes less than a certain predetermined value to ensure that their frequency response is identical to the maximum extent (an illustrative example is shown in Fig. 1).

Signals that have passed the low-pass filter and the band-pass filter are subtracted from one another. The received signals are converted to digital form in the corresponding analog-to-digital converters (ADCs). These values are summarized and stored. From the received amounts, the sum with the maximum value is found. The threshold value is determined by multiplying the found maximum value by a coefficient, the value of which is set in advance.

The value of this coefficient is determined at the development stage experimentally or by mathematical modeling as the value that provides the maximum probability of the correct detection of signals, provided that the level of false alarm, i.e. making a decision on the presence of a signal in its absence does not exceed a predetermined level.

The obtained values of the sums are compared with a threshold; according to the results of the comparison, a conclusion is made about the presence or absence of signals with an appropriate frequency.

The process of detecting a multi-frequency signal using modulation by frequency shift and compensation of combination components in the presence of interference such as additive white Gaussian noise (ABGS) is simulated.

The interference in the simulation is presented in the form of additive white Gaussian noise, i.e. a set of harmonic oscillations with random values of amplitudes (U _pi ) and phases (ϕ _pi ), which are distributed according to normal (amplitudes) and uniform (phases) laws (see, for example, the study guide “Fundamentals of the theory of radio engineering systems.” / V.I. Borisov, V.M. Zinchuk, A.E. Limarev, N.P. Mukhin, Edited by V.I. Borisov, Voronezh Telecommunication Research Institute, 2004., p. 51)

, (6)U =

, (6)

– частота, фаза и амплитуда i-ой составляющей помехи, соответственно;where: ω _pi , φ _pi ,

- frequency, phase and amplitude of the i-th component of the interference, respectively;

Nsp is the number of harmonic noise components used to represent it.

The frequencies of the interference components were modeled as random variables whose values are distributed according to a uniform law in the signal band.

Noise samples are independent random variables.

The results of evaluating the effectiveness of the proposed method were obtained by mathematical modeling on a computer using the MATLAB system.

When modeling, the following initial data were used:

- the number of implementations - 1000;

- the number of interference components - 1000;

- the number of harmonic signals - 8;

- values of harmonics frequencies (in arbitrary units): 10.0; 10.1; 10.2; 10.3; 10.4; 10.5; 10.6; 10.7;

- the amplitude of the harmonic signals is 1;

- the number of samples for the period - 2;

- the number of periods - 5;

- the amplitude of the interference is 26.0;

- the threshold value for the amplitude of the signals is 4.9;

- sampling rate –1;

- compensation coefficient of the combination components in the frequency region where the frequency response of the bandpass filter is close to the frequency response of the low-pass filter - 0.05;

- the compensation coefficient of the combination components in the frequency region close to zero, where the slope of the frequency response of the bandpass filter is maximum, is calculated provided that in this case the frequency response of the bandpass filter has a linear relationship.

The simulation results of the decision process on the presence of a signal for the proposed method is for the ratio of the interference power to the signal power 7 and the probability of false alarm 10 ^{-3, the} probability of making the right decision on the presence of a signal is at least 0.999 for each frequency component (subcarrier) of the signal.

When modeling the prototype method for signals with OFDM, the following values of harmonic frequencies (in arbitrary units) were established: 1; 2; 3; 4; 5; 6; 7; 8.

Based on the simulation results, it was found that for the prototype method, the probability of a false alarm equal to 10 ^-3 , the probability of making the right decision about the presence of a signal equal to 0.999, is provided when the ratio of interference power and signal is equal to 1.

That is, the effectiveness of the proposed method in terms of the ratio of the noise and signal powers exceeds the efficiency of the prototype method by almost 7 times. In this case, the ratio of the signal bandwidth for the method under consideration (0.7) to the signal band used for the prototype method (7) is 0.1. That is, the sensitivity of the receiver of the communication medium in which the proposed method is implemented is 10 times lower than the sensitivity of the receiver of the communication medium in which the prototype method is implemented. Thus, the effectiveness of the proposed method in terms of the ratio of the noise and signal powers exceeds the efficiency of the prototype method by almost 70 times.

From the fact that for the implementation of the proposed method it is enough to use two samples per period, and for the prototype method no less than 10 - 15 samples per period and the fact that the value of the upper frequency of the signal spectrum with OFDM significantly exceeds the value of the upper frequency of the signal spectrum used in the proposed method, it follows that the rate of exchange of information when using the proposed method significantly exceeds the rate of exchange of information that can be achieved using the prototype method.

The structural diagram of a device that implements the proposed method is shown in FIG. 2, where indicated:

1.1 - 1.n - multiplication blocks from the first to the n-th;

2.1 - 2.n - low-pass filters (low-pass filters) from the first to the n-th;

3.1 - 3.n - subtraction devices from the first to the n-th;

4.1 - 4.n - analog-to-digital converters (ADCs) from the first to the n-th;

5.1 - 5.n - band pass filters from the first to the n-th;

6 - computing device (WU).

The device contains n parallel rulers, each of which consists of a corresponding series-connected multiplication unit 1, low-pass filter 2, a subtraction device 3 and an ADC 4, while a band-pass filter 5 is connected between the output of the multiplication unit 1 and the second input of the subtraction device 3. Inputs of n multiplication blocks 1.1 - 1.n are combined and are the input of the device. The outputs n of the ADC 4.1 - 4.n are connected to the corresponding inputs of the computing device 6, the output of which is the output of the device. In this case, the second inputs of the multiplication blocks 1.1 - 1.n are inputs for the reference signals.

The device operates as follows.

The signals are formed as the sum of n harmonic signals (subcarriers) with different frequencies using frequency shift modulation. The values of neighboring frequencies differ by a certain value Δf _ij .

Here i, j are the numbers of neighboring frequencies, j = i + 1.

The values of the frequency shifts are set so that the difference value of any pair of frequencies does not exceed the frequency value for which the difference in the amplitude-frequency characteristic (AFC) of the low-pass filter 2 and the band-pass filter 5 becomes less than some predetermined value (F _p ) (see Fig. 1) .

The reference frequencies are formed with the same values as the harmonic signals.

The number of harmonic components n used in the formation of the signal, the values of the frequency shifts between the signals are determined at the development stage by experiment or by mathematical modeling as values that provide the maximum degree of noise immunity at a given level of data exchange rate.

At the stage of entering into communication, the signal is synchronized by any known method, for example, by processing the code sequence used.

The adopted additive mixture of signal and interference is supplied to the first inputs of the multiplication blocks 1.1-1.n, to the second inputs of which they supply the corresponding reference signals.

The result of multiplying the signal and interference by reference signals branch into two identical components. The first component is filtered by the low-pass filter 2.1 - 2.n, the band of each of which is consistent with the signal band. At the same time, the second component is filtered with band-pass filters 5.1 - 5.n, the passband of each of which is selected so that the upper frequency of the band-pass filters 5.1 - 5.n corresponds to the upper frequency of the signal, the lower frequency of the band-pass filter is set as close to zero as possible.

The selection of the low-pass filter 2.1 - 2.n and the band-pass filters 5.1 - 5.n is carried out with phase-frequency characteristics identical to the maximum and so that the frequency response of the band-pass filters in the frequency region close to zero has the maximum possible slope in the frequency region, starting from values for which the difference between the frequency response of the low-pass filter 2.1 - 2.n and the band-pass filters 5.1 - 5.n becomes less than some predetermined value, ensure that their frequency response is identical to the maximum extent (an illustrative example is shown in Fig. 1).

Signals that pass through the low-pass filter 2.1 - 2.n and bandpass filters 5.1 - 5.n, subtract one from the other, respectively. That is, the signal of the first bandpass filter 5.1 is subtracted from the signal of the first LPF 2.1, the signal of the second bandpass filter 5.2 is subtracted from the signal of the second LPF 2.2, etc.

The received signals are converted to digital form in the corresponding ADC 4.1 - 4.n. These signals are digitally supplied to the computing device 6.

In the computing device 6, these values are summarized and stored. From the received amounts, the sum with the maximum value is found. The threshold value is determined by multiplying the found maximum value by a coefficient, the value of which is set in advance.

The value of this coefficient is determined at the development stage experimentally or by mathematical modeling as the value that provides the maximum probability of the correct detection of signals, provided that the level of false alarm, i.e. making a decision on the presence of a signal in its absence does not exceed a predetermined level.

The obtained values of the sums are compared with a threshold; according to the results of the comparison, a conclusion is made about the presence or absence of signals with an appropriate frequency.

The simulation results of the multi-frequency signal detection process using frequency shift modulation and compensation of combination components in the presence of interference such as additive white Gaussian noise (ABGS) are given above.

Multiplication blocks 1.1 - 1.n can be made, for example, in the form of a frequency converter (mixer), see, for example, the training manual "Fundamentals of the theory of radio systems". Tutorial. //IN AND. Borisov, V.M. Zinchuk, A.E. Limarev, N.P. Mukhin. Ed. IN AND. Borisov. Voronezh Scientific Research Institute of Communications, 2004 ”, pp. 186 - 189.

ADC 4.1 - 4.n can be performed, for example, on the AD7495BR chip from Analog Devices.

Computing device 6 can be performed, for example, as a single microprocessor device with appropriate software, for example, a processor series TMS320VC5416 from Texas Instruments, or as a programmable logic integrated circuit (FPGA), with appropriate software, for example, Xilinx FPGA from Xilinx.

Thus, the inventive method can be implemented by the described device.

Claims (1)

A method of isolating a signal using frequency shift modulation and compensation of combinational components, which consists in generating a signal consisting of several harmonic signals using frequency shift modulation (FSK), at the stage of entering into communication, the communication means are synchronized, after multiplying by the corresponding reference signals S _op.i in the multiplication units, the result of signal and noise conversion is formed, i.e. multiplication by reference signals, characterized in that the frequency shifts between adjacent signals are set in advance so that the difference value of any pair of frequencies does not exceed the frequency value for which the difference in the amplitude-frequency characteristic (AFC) of the low-pass filter (LPF) and the band-pass filter becomes less than a predetermined value; Signal processing is carried out simultaneously and in the same way in the respective lines, namely: after multiplying by the corresponding reference signals, the results of signal conversion and interference are generated in the multiplication units, each of which is branched into two identical components: the first component is filtered by the low-pass filter, the band of which is matched with the signal band ; at the same time, the second component is filtered by a band-pass filter, the passband of which is selected as follows: the upper frequency of the band-pass filter corresponds to the upper frequency of the signal; the lower frequency of the band-pass filter is set as close to zero as possible; the low-pass filter and the band-pass filter are selected with phase-frequency characteristics identical to the maximum and so that the frequency response of the band-pass filter in the region of frequencies close to zero has the maximum slope; in the frequency range, starting from a value for which the difference between the frequency response of the low-pass filter and the band-pass filter becomes less than a certain predetermined value, they ensure that their frequency response is identical to the maximum, the signals that have passed the low-pass filter and the band-pass filter are subtracted from each other in each line, the received signals converted to digital form in the corresponding ADC; the obtained values are summarized and stored; from the received amounts, the sum with the maximum value is found; determining a threshold value by multiplying the found maximum value by a coefficient whose value is set in advance; the obtained values of the sums are compared with a threshold; according to the results of the comparison, a conclusion is made about the presence or absence of a signal with an appropriate frequency.

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