# The method of adjustment of the frequency coefficients of the transmission channel multi-channel receiver

The invention relates to the field of adaptive antennas and systems navigation. The essence of the invention is that the method contains a reference channel and adjust the channel filter, which consists in the fact that the input signal to adjust the channel calculates the filter coefficients and adjust CCP required under CCP, characterized in that the reference and adjust the channels originally used the same digital filters with finite impulse response (FIR) on the reference inputs and adjust channels serves the input signal in the form of test photomanipulating signal with a leap phaseradian, which measure transient response (HRP) at the output CP of the reference channel and PH at the inlet fit adjustable channel, and the measured PH compute the coefficients of the fit FIR adjust channel by the following formulas: W=(XH)^{-1}Y,orwhere W is a column vector of desired weight multipliers to find the vector-column of coefficients fit FIRN - diagonal matrix coecients of the original ZF SUCH THAT X is the matrix of the samples at the input of ZF SUCH Polstra hemy when implementing the method, is to increase the depth of suppression. 16 Il.

1. The technical field to which the invention relates

Adaptive antennas, radio navigation system.

2. The level of technology

The prior art can be identified by the following sources.

1. Kartashevskaya street Century, the Processing of spatio-temporal signals in channels with memory. - M.: Radio and communication, 2000. - 272 S. [1]. We consider the problem of estimating the impulse response in the channel memory in the transmission systems of discrete messages. The proposed algorithms for estimating impulse response on the ideal classified sample and conditionally classified the sample, including the method of regularization. Matching significant feature is that is determined by the response of the channel for a test or working signal. The difference is at the source [1] is estimated impulse response, and in our case, the measured transient response.

The drawback at the source [1] - it is difficult to determine the impulse response, the MSE of the estimates of the impulse response depends on the delay with respect to the quantum in the moments determine the impulse response. Using a complex algorithm demodulation.

2. Monzingo R. A., Miller, T. C. Adaptive antenna recaptives canceller of interference on multi-tap delay line (MLA), proposed for use in a broadband communication line. Matching essential features: multichannel adaptive interference canceller at the MLA allows you to compensate for channel mismatch with the right choice of the number of taps of the delay line; calculated material weight coefficients for the transversal filter. The difference is the tuning frequency of the transmission ratios (CCP) channels of the source [2] is carried out in the adaptation algorithm minimum mean square error on the intermediate frequency directly, i.e., in analog form.

Disadvantages: only works on adaptive procedure, therefore, the settling time is unknown; the number of taps of the filter is calculated approximately; it is difficult to make real device analog delay elements and require a large setup time and debugging. The closest analogue (prototype) the author believes transversal filter as part of the interference canceller.

3. The invention

Consider the M-channel receiving satellite communication system with spatially separated (see Fig.1). The main elements of the receive path are antenna (1); microwave tract (MW tract, 2); path p is logo-digital Converter (ADC, 5); digital filter (ZF, 6); a digital processor (CPU 7). The sampling rate of the ADC and the filter fit in much more interest exceeds the upper frequency.

In theory, the communication channels describes the impulse response (in the time domain or the frequency transfer coefficient (frequency). If the input channel to apply a stepped effect on the output you will get transient response (HRP). The impulse response is derived from the transitive characteristics, i.e., it determines uniquely. Therefore, the channels will be identical in case of equality in any of these characteristics.

If the adjustment is done in order to obtain identical channels in a multichannel receiver, the available channels, you select one, let's call it a benchmark by which to adjust the rest. Other variants of adjustment when an arbitrarily chosen reference channel adjusts at least one channel of the multichannel receiver. Adjustment of the frequency coefficients of the transmission channel is performed using the adjustment coefficients fit thus to adjust the channel to obtain a PH equal to the PH of the reference channel.

Synthesis of CP will perform based on the calculation of transversal the factors fit is calculated for each quadrature channel separately (as hereinafter defined for one channel). Structural diagram of the digital FIR filter shown in Fig.2. The FIR filter consists of elements of the delay registers (8), multipliers on the coefficients of (9) and adder (10). Let us introduce the notation:

x(n) is the digitized signal at the ADC output, n is the number of the reference in discrete time;

h(n) is the discrete impulse response (DIC) digital filter;

weighted DICH digital filter, so that

where w(n) are weighting factors.

y(n) - output process at the output of the digital filter.

It is known that the process at the output of the digital filter is a discrete convolution of the input process with DIG:

N - order FIR filter.

The order of the FIR filter is selected such that the total time delay elements of the delay was not shorter than the duration of HRP (at the output of this filter) determining the acceptable level of residual misalignment of the channels.

In the method of adjusting the frequency coefficients of the transmission channel to obtain the transient response to the inputs of these channels serve a probationary impact in the form of a phase jumpradian photomanipulating signal with a duration in excess of not less than 3-5 R is lnost initial state before the jump is the same and measure PH: (n) at the output CP of the reference channel, x(n) input adjustable fit channel. The PH measurement shall be performed with synchronous (frequency or phase) of the reference signal generators in the reference and adjust channels: microwave and inverter circuits (2, 3 of Fig.1) frequency total variation not exceeding 5% of bandwidth ZF (-3 dB), generators receive the I and q channels videocassette exactly in phase (up to 5), generators, frequency F_{D}sampling the signals on videocassette or transfer to videocasette exactly in phase (up to 5), generators, frequency F_{D}sampling the signals at the if with transfer to the inverter variation not exceeding 5% of bandwidth ZF (-3 dB), digital generators accurately clock and values with an accuracy of up to 0.5 lower significant bits of the transfer of the spectrum at zero frequency immediately after the ADC (if all these generators are used). To achieve acceptable results, the signal-to-noise ratio in the measured channel must be at least 20 dB. After measurements replace HRP output fits on the adjustable channel at the appropriate PH(n) of the reference channel and using HRP x(n) at the input fits solve the convolution equation for noorden is blowing this:

- a column vector of the desired transient response (HRP reference channel),^{T}the symbol of the transpose of the vector;

where N is the diagonal matrix coecients of the original FIR filter.

where X is the matrix of the samples at the ADC output (adjustable channel).

where W is a column vector of desired weight multipliers (is defined similarly).

HN - matrix equivalent discrete convolution. After solving the equation (2) found coefficients of the FIR filter are determined by the formula:

At a PH Y of the reference channel and HRP X adjust channel may be finding the coefficients of the FIR filter:

Write equations (2) and (3) possible in an equivalent form for the transposed vectors and matrices.

The solution of the matrix equation (2) with respect to W can be made in any known manner, for example according to the method of treatment of the matrix HN with regularization. The solution of the matrix equation (3) with respect toAfter installation found by this method factors in ZF adjust channel automatic frequency approximation of the coefficient of transmission adaptive jitter channel-to-frequency ratio of the reference channel.

The method of adjustment CCP channels is used on the output of the ADC channels are initially identical fits directly involved in filtering, and digital processor which carries out the measurement of HRP channels and the calculation of the coefficients of the fits tweakable channels.

The technical result. The method of adjustment allows you to receive channels with reduced mutual differences in CCP channels, which reduces the requirements on the precision of previous analog filters and other devices. Thereby achieve cost savings and improved performance of the receiving system. Reduction of non-identical channels, resulting from the adjustment provides an increase in the depth of suppression in adaptive antenna arrays and the interference compensators. The use of channels with the same frequency response and the phase response curve allows for more accurate opredelit the Central frequency of the spectrum allows to reduce the requirements for device synchronization. If the reference channel to take the channel without inter-symbol distortion (ISI), then this method it is also possible to reduce ISI in an adjusted reference channel channels.

4. List of figures

Fig.1. Structural scheme of the M-channel receive path of a satellite communications system with spatially separated.

Shows the major nodes of the receiving station sputnikovoi communication. Shows the location of the digital filter (6) and a digital processor (7).

Fig.2. Structural diagram of the digital FIR filter.

Depicts a transversal structure with elements of delay (denoted by T), multipliers signals from the outputs of the delay elements on the coefficientsand adder.

Fig.3. Model M-canaliega receive path.

The equivalent model of the circuit shown in Fig.1. Shows the scheme on the basis of which the mathematical model in Matlab F. "MathWorks, Inc".

Fig.4. AFC 0-th channel.

Tuning on the bass. The signal-to-noise ratio AF 30 dB.

Fig.5. Response 0-th channel.

Tuning on the bass. The signal-to-noise ratio AF 30 dB.

Fig.6. The initial response of the M-th channel.

Tuning on the bass. The signal-to-noise ratio AF 30 dB.

Fig.7. The initial response of the M-th channel

Tuning on the bass. Signal/showhide AF 30 dB.

Fig.9. Adjusted response of the M-th channel.

Adjusting the fit of the lower frequencies. The signal-to-noise ratio AF 30 dB.

Fig.10. AFC 0-th channel.

The adjustment of the inverter. The signal-to-noise ratio AF 40 dB.

Fig.11. FR 0 th kakala.

The adjustment of the inverter. The signal-to-noise ratio AF 40 dB.

Fig.12. The initial response of the M-th channel.

The adjustment of the inverter. The signal-to-noise ratio AF 40 dB.

Fig.13. The initial response of the M-th channel

The adjustment of the inverter. The signal-to-noise ratio AF 40 dB.

Fig.14. Adjusted frequency response of the M-th channel.

The tunable bandpass fit. The signal-to-noise ratio AF 40 dB.

Fig.15. Adjusted response of the M-th channel.

The tunable bandpass fit. The signal-to-noise ratio AF 40 dB.

Fig.16. Structural diagram of the practical implementation of digital FIR-filter.

Shown saving scheme for implementation on FPGA.

5. Information confirming the possibility of carrying out the invention

In tlab simulated equivalent part of the receiving system shown in Fig.1, starting from the analog filter AF (4) and before the release of ZF (6). The noise at the input fits to model discrete additive white Gaussian noise (DBGS). The model diagram shown in Fig.3.

Via the antenna (1) signal goes in the microwave (2) and FC (3) tracts, where it is transferred to promezhutochny signal is subjected to sampling and quantization, also you can transfer videocasette. After the ADC signal passes through ZF (6), with an adjusted odds to reduce the non-identical channels 0 and M. PH at the entrance of fit is measured CPU (7).

In Matlab speed impact was simulated leap phaseradian photomanipulating signal (FM) with initial phase, corresponding to the maximum response. The sampling rate in the model is 12 MHz.

The simulation results.

1. Tuning on the bass.

Fits in the form of a FIR lowpass filter 24, the cutoff frequency F_{with}-3 dB is 1.2 MHz, the frequency F_{C}the beginning of the detention area on level -40 dB and 2.1 MHz. Irregularity in the band of 0.1 dB. The capacity factors of 12 bits.

AF is modeled filter Chebyshev-1 bass 3 order, F_{with}2.4 MHz, F_{C}- 4.35 MHz for channel 0 and F_{with}equal to 1.25 MHz, F_{C}- 3.0 MHz for M-channel non-uniformity in the band 3 dB for both channels. The ADC is 10 bits. The regularization parameterin the formula for matrix inversion (HN+I)^{-1}equal to 10^{-9}.

The simulation results shown in Fig.4-9 for the signal-to-noise ratio AF of 30 dB. AFC and PFC are using on the toty slice F_{c}-3 dB is 2.0 and 4.9 MHz, frequency F_{C}detention area at the level of -37 dB - 1,65 and 5.3 MHz. Irregularity in the band of 0.1 dB. The capacity factors of 14 bits.

AF is simulated band-pass filter Chebyshev-1 6 order, F_{with}are 2 and 5 MHz by non-uniformity in the band - 0.1 dB for channel 0 and 2.25 and 4,75 MHz when the unevenness of 3 dB for M-th channel. ADC - 12-bit. The regularization parameterin the formula for matrix inversion (HN+I)^{-1}equal to 10^{-3}.

The simulation results shown in Fig.10-15 for the signal-to-noise ratio AF 40 dB.

AFC and PFC are using the discrete Fourier transform.

Conclusions from the simulation.

For the treatment of matrices X and XH, composed of samples x_{-(N-1)}..x_{0}..x_{N-1}it is necessary for the response to the test signal X was present in about half of the samples: x_{0}..x_{N-1}.

Day best measurement for calculating the coefficients of the fit are recommended with a minimum of noise.

The practical realization of FIR filters.

In Fig.16 shows a structural diagram of a digital FIR filter 24 procedure for its implementation on FPGA. This scheme allows the use of smaller the RA (10) and adder reset (11), components of a high-speed arithmetical-logical unit (ALU). The source data for the ALU comes from RAM 1-4 (12). The data in the RAM 1-2 cyclically updated. On the RAM 2, the data are delayed by the delay time in the RAM 1, which is illustrated by the delay element (13). RAM 3-4 are the filter coefficients can be updated in real-time. For generating addresses for reading RAM 3-4 required counter 1 (14). For generating addresses for reading the RAM 1-2 necessary counter 2 (15). Counter 3 (16) provides the address of the bootstrap for counter 2. If RAM 1-2 populated data with ascending numbers, address RAM 3-4 see and read the coefficientsalso with increasing addresses, the counter 2 is subtractive.

For the implementation of FIR filters large orders of magnitude diagram in Fig.14 can be generalized. For this purpose it is necessary to repeat elements (12, 13, 9, 10) the required number of times that depends on the order of filter and frequency calculations, and to combine the common adder (11).

Claims

The method of adjusting the frequency of transmission ratios (CCP) multi-channel receiver, containing the reference channel and adjust the channel filter, zakljucaju required under CCP, characterized in that the reference and adjust the channels originally used the same digital filters with finite impulse response (FIR) on the reference inputs and adjust channels serves the input signal in the form of test photomanipulating signal with a leap phaseradian, which measure transient response (HRP) at the output CP of the reference channel and PH at the inlet fit adjustable channel, and the measured PH compute the coefficients of the fit FIR adjust channel by the following formulas:

W=(XH)^{-l}Y,or

where W is a column vector of desired weight multipliers to find the vector-column of coefficients fit FIR;

N - diagonal matrix coecients of the original ZF FIR;

X - matrix samples at the input fits FIR adjust channel;

Y is a column vector of counts of HRP at the output of ZF SUCH reference channel.

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