CN114844553B - Single symbol rate sampling method based on prior filtering for high-speed transmission - Google Patents

Abstract

The utility model provides a single code element rate sampling method based on prior filtering for high-speed transmission, which is applied to the technical field of satellite communication and signal processing, wherein, the receiver device is satellite device or ground device, and the receiver device comprises an analog-to-digital conversion chip and a digital processing chip. The analog-to-digital conversion chip processes the received time domain continuous signal to obtain a first sampling signal; the first sampled signal corresponds to a first sampling frequency, the first sampling frequency being a single symbol rate; the analog-to-digital conversion chip sends the first sampling signal to the digital processing chip; the digital processing chip calculates an interpolation value at each interpolation time according to the first sampling signal, and combines the first sampling signal and the interpolation value into a second sampling signal according to the interpolation time; and the second sampling frequency corresponding to the second sampling signal is 2 times of the code element rate. The method can reduce the requirement on the sampling rate of the analog-digital conversion chip and can reduce the power consumption of the analog-digital conversion chip.

Description

Single symbol rate sampling method based on prior filtering for high-speed transmission

technical field

The present disclosure relates to the technical field of satellite communication and signal processing, and in particular to a single symbol rate sampling method based on prior filtering applied to high-speed transmission.

Background technique

The Nyquist sampling theorem is one of the basic laws that need to be observed in the field of communication and signal processing. The Nyquist sampling theorem includes: the sampling frequency f _s is greater than twice the highest frequency f _max corresponding to the time-domain signal, that is, when f _s >=2f _max , the discrete signal after sampling can completely retain the information in the time-domain signal, and will not cause spectral aliasing. When the sampling frequency does not satisfy the Nyquist sampling theorem, spectrum aliasing will occur, affecting subsequent functions such as channel compensation, spectrum display, and clock recovery.

When the symbol rate to be achieved is high, the inability to obtain a suitable analog-to-digital conversion chip is a technical problem that needs to be solved.

Contents of the invention

In order to overcome the problems existing in related technologies, the present disclosure provides a priori filtering-based single symbol rate sampling method applied to high-speed transmission, which is applied to receiver equipment, and is applied to receiver equipment. The receiver equipment is satellite equipment Or ground equipment, the receiver equipment includes an analog-to-digital conversion chip and a digital processing chip, including:

The analog-to-digital conversion chip processes the received time-domain continuous signal to obtain a first sampling signal; wherein, the first sampling signal corresponds to a first sampling frequency, and the first sampling frequency is a single symbol rate;

The analog-to-digital conversion chip sends the first sampling signal to a digital processing chip;

The digital processing chip calculates an interpolation value at each interpolation moment according to the first sampling signal, and combines the first sampling signal and the interpolation value into a second sampling signal according to the interpolation moment; wherein, the The second sampling signal corresponds to a second sampling frequency, and the second sampling frequency is twice the symbol rate.

In one embodiment, the method also includes:

The sending device uses M times the first sampling frequency to sample the time-domain continuous signal to obtain a third sampling signal; wherein, the M is an integer greater than 1; wherein, the first sampling frequency is a single code meta rate;

Set M-1 sampling points at the same position in the consecutive M sampling points corresponding to each symbol in the third sampling signal to zero to obtain a fourth sampling signal;

and sending a time-domain continuous signal corresponding to the fourth sampling signal.

In an embodiment, the analog-to-digital conversion chip processes the received time-domain continuous signal to obtain a first sampling signal, including: the analog-to-digital conversion chip uses the first sampling frequency to process the time-domain continuous signal The signal is sampled to obtain a first sampling signal; wherein, the first sampling frequency is a single symbol rate.

In an implementation manner, the interpolation time is a zero-crossing position between corresponding peak points of two adjacent symbols in the first sampling signal.

In an embodiment, the interpolation value at the interpolation time is a superposition value at the interpolation time of the time-domain signals after the impact signals at all sampling time in the first sampling signal pass through a band-pass filter.

In one embodiment, the method also includes:

Using a test signal known to both the receiver device and the sender device, to test the spectral response of the target channel, and determine the spectral characteristics of the bandpass filter corresponding to the target channel;

Wherein, the test signal includes all spectral components in the band of the target channel; the target channel includes a shaping filter of the sending device, a wireless channel for transmitting wireless data, and a matched filter of the receiving device, and the target channel is the the linear time-invariant channel.

In one embodiment, the system function of the bandpass filter is a raised cosine function, the shaping coefficient is 0.5, and the corresponding number of symbols is greater than 30.

In one embodiment, the number of symbols is 31.

The technical solutions provided by the embodiments of the present disclosure may include the following beneficial effects:

1. The signal sampling frequency is reduced, thereby reducing the demand for the sampling rate of the analog-to-digital conversion chip, which is conducive to the realization of ultra-high-speed signal sampling processing.

2. The power consumption of the analog-to-digital conversion chip is reduced.

3. The signal transmission rate of the analog-to-digital conversion chip and the digital processing chip is reduced, and the problem of high-speed digital interface is solved.

It is to be understood that both the foregoing general description and the following detailed description are exemplary and explanatory only and are not restrictive of the present disclosure.

Description of drawings

The accompanying drawings, which are incorporated in and constitute a part of this specification, illustrate embodiments consistent with the invention and together with the description serve to explain the principles of the invention.

Fig. 1 is a flowchart of a single symbol rate sampling method based on a priori filtering applied to high-speed transmission according to an exemplary embodiment;

FIG. 2 is a flow chart of another priori filtering-based single symbol rate sampling method applied to high-speed transmission according to an exemplary embodiment;

Fig. 3 is a schematic diagram showing time-domain extension of an impulse signal after being band-limited according to an exemplary embodiment.

Detailed ways

Reference will now be made in detail to the exemplary embodiments, examples of which are illustrated in the accompanying drawings. When the following description refers to the accompanying drawings, the same numerals in different drawings refer to the same or similar elements unless otherwise indicated. The implementations described in the following exemplary examples do not represent all implementations consistent with the present invention. Rather, they are merely examples of apparatuses and methods consistent with aspects of the invention as recited in the appended claims.

Applicants have found in their research that:

1. When the symbol rate is high, the receiving device requires an analog-to-digital conversion chip with a high sampling rate, especially when the symbol rate is high to a certain extent, there may be no suitable analog-to-digital conversion chip to meet the usage requirements.

2. When the symbol rate to be achieved is high, after the analog-to-digital conversion chip of the receiving device sends the sampled data to the digital processing chip, the data transmission rate between the analog-to-digital conversion chip and the digital processing chip is high, resulting in Huge data transmission pressure.

Embodiments of the present disclosure provide a single symbol rate sampling method based on a priori filtering that is applied to high-speed transmission. Applied to a receiver device, which is a satellite device or a terrestrial device. The receiver device includes an analog-to-digital conversion chip and a digital processing chip.

As shown in Figure 1, the prior filtering-based single symbol rate sampling method applied to high-speed transmission includes:

Step S101, the analog-to-digital conversion chip processes the received time-domain continuous signal to obtain a first sampling signal; wherein, the first sampling signal corresponds to a first sampling frequency, and the first sampling frequency is a single symbol rate;

Step S102, the analog-to-digital conversion chip sends the first sampling signal to a digital processing chip;

Step S103, the digital processing chip calculates an interpolation value at each interpolation time according to the first sampling signal, and combines the first sampling signal and the interpolation value into a second sampling signal according to the interpolation time; Wherein, the second sampling signal corresponds to a second sampling frequency, and the second sampling frequency is twice the symbol rate.

In one implementation manner, the single symbol rate in step S101 is the highest frequency corresponding to the time-domain continuous signal. The double symbol rate is twice the highest frequency corresponding to the time-domain continuous signal.

Embodiments of the present disclosure can achieve the following technical effects:

1. The signal sampling frequency is reduced, thereby reducing the demand for the sampling rate of the analog-to-digital conversion chip, which is conducive to the realization of ultra-high-speed signal sampling processing.

2. The power consumption of the analog-to-digital conversion chip is reduced.

3. The signal transmission rate of the analog-to-digital conversion chip and the digital processing chip is reduced, and the problem of high-speed digital interface is solved.

Embodiments of the present disclosure provide a single symbol rate sampling method based on a priori filtering that is applied to high-speed transmission. Applied to a receiver device, the receiver device is a satellite device or a ground device. The receiver device includes an analog-to-digital conversion chip and a digital processing chip.

The method includes step S101 to step S103 described in step S100-1.

Wherein, step S100-1 specifically includes:

The sending device uses M times the first sampling frequency to sample the time-domain continuous signal to obtain a third sampling signal; wherein, the M is an integer greater than 1; wherein, the first sampling frequency is a single code meta rate;

In order that the signal can be transmitted in a bandwidth-limited channel, the transmission signal needs to be whitened, and only one sampling point is reserved for each symbol, and the other sampling points are set to zero values, that is: each of the third sampling signals is M-1 sampling points at the same position in the consecutive M sampling points corresponding to the symbol are set to zero to obtain a fourth sampling signal;

and sending a time-domain continuous signal corresponding to the fourth sampling signal.

In one example, the value of M is 3. In step S201, each symbol in the third sampling signal corresponds to 3 consecutive sampling points. Wherein, setting the M-1 sampling points at the same position in the consecutive M sampling points corresponding to each symbol in the third sampling signal to zero means:

Set the value of the first 2 sampling points in the 3 consecutive sampling points corresponding to each symbol in the third sampling signal to 0,

Or, set the value of the last 2 sampling points in the 3 consecutive sampling points corresponding to each symbol in the third sampling signal to 0,

Or, set the values of the first sampling point and the third sampling point among the three consecutive sampling points corresponding to each symbol in the third sampling signal to 0.

In one example, the value of M is 4. In step S201, each symbol in the third sampling signal corresponds to 4 consecutive sampling points. Wherein, setting the M-1 sampling points at the same position in the consecutive M sampling points corresponding to each symbol in the third sampling signal to zero means:

Set the value of the first 3 sampling points in the 4 consecutive sampling points corresponding to each symbol in the third sampling signal to 0,

Or, set the values of the last 3 sampling points in the 4 consecutive sampling points corresponding to each symbol in the third sampling signal to 0,

Or, setting the values of the other 3 sampling points in the 4 consecutive sampling points corresponding to each symbol in the third sampling signal except the 2nd sampling point are 0;

Or, set values of the other three sampling points except the third sampling point among the four consecutive sampling points corresponding to each symbol in the third sampling signal to be 0.

Embodiments of the present disclosure provide a priori filtering-based single symbol rate sampling method applied to high-speed transmission. Applied to a receiver device, the receiver device is a satellite device or a ground device. The receiver device includes an analog-to-digital conversion chip and a digital processing chip.

This method includes the steps S101 to S103.

Wherein, the analog-to-digital conversion chip described in step S101 processes the received time-domain continuous signal to obtain the first sampling signal, including:

The analog-to-digital conversion chip uses the first sampling frequency to sample the time-domain continuous signal to obtain a first sampling signal; wherein, the first sampling frequency is a single symbol rate.

Embodiments of the present disclosure provide a priori filtering-based single symbol rate sampling method applied to high-speed transmission. Applied to a receiver device, the receiver device is a satellite device or a ground device. The receiver device includes an analog-to-digital conversion chip and a digital processing chip.

This method includes the steps S101 to S103.

In addition, the interpolation time is a zero-crossing position between corresponding peak points of two adjacent symbols in the first sampling signal.

The interpolation value at the interpolation moment is a superposition value at the interpolation moment of time-domain signals after the impact signals at all sampling moments in the first sampling signal pass through a band-pass filter.

In the embodiment of the present disclosure, by using the superimposed value as the interpolation value at the interpolation time, the first side lobe of the spectrum of the interpolation point can be suppressed, and the noise value of the interpolation point can be reduced.

Embodiments of the present disclosure provide a priori filtering-based single symbol rate sampling method applied to high-speed transmission. Applied to a receiver device, the receiver device is a satellite device or a ground device. The receiver device includes an analog-to-digital conversion chip and a digital processing chip.

As shown in FIG. 2 , the priori filtering-based single symbol rate sampling method applied to high-speed transmission includes: step S100 and the steps S101 to S103.

Wherein, step S100 includes: using a test signal known to both the receiving device and the sending device, to test the spectral response of the target channel, and determine the spectral characteristics of the bandpass filter corresponding to the target channel.

Wherein, the test signal includes all spectral components in the band of the target channel; the target channel includes a shaping filter of the sending device, a wireless channel for transmitting wireless data, and a matched filter of the receiving device, and the target channel is the the linear time-invariant channel.

A specific example will be described below.

specific embodiment

Take the scenario of transmitting data from satellite remote sensing data to the ground as an example.

，且符合奈奎斯特无码间串扰的条件，其时 域的系统函数为h(t) 。 The test signal is used to test the target channel to determine the spectrum characteristics of the band-pass filter corresponding to the target channel. Determine the frequency domain system function of this bandpass filter as

, and meet the Nyquist condition of no intersymbol interference, the system function in the time domain is h(t) .

During the orthogonal transmission of signals, parallel processing is performed on the I-channel and Q-channel signals. For ease of expression, only the processing process for the I-channel signal is described here.

Before transmission, the expression of channel I signal is s(t) , as shown in formula (1):

(1)

(1)

Where T is the symbol period, a(t) is a random variable carrying information, and its amplitude value changes with the baseband information and the modulation mapping relationship. Under the quadrature phase shift keying (Quadrature Phase Shift Keying, QPSK) system , the value of a (t) is shown in formula (2):

(2)

(2)

： The FPGA of the satellite load samples s(t) M times, and the sampled signal is shown in formula (3)

:

(3)

(3)

Among them, δ(t) is the impact signal; m represents the sampling point number of each symbol, and the value ranges from 0 to M-1; T _S is the sampling period.

In order for the signal to be transmitted in the bandwidth-limited channel, the transmission signal is whitened, that is, only one sampling point is reserved for each symbol, and other sampling points are changed to zero values.

Define the value of s(t) in the time interval [ nT,(n+1)T ] as a _n , as shown in formula (4):

(4)

(4)

Among them, a _n is a random variable, which takes a constant value in each symbol period.

The expression after whitening processing is shown in formula (5):

(5)

(5)

通过信道

后，频域带宽受到切割，信号在时域内进行扩展，如图3 所示。 Signal

through the channel

After that, the frequency domain bandwidth is cut, and the signal is expanded in the time domain, as shown in Figure 3.

The time-domain expression of the continuation of the impulse signal after passing through the band-limited system is the impulse response of the system, where the system function of the channel and the impulse response are a pair of Fourier transform pairs.

的第N个码元的冲击样点值为

The impact sample point value of the Nth symbol is

(6)

(6)

The time-domain signal of the shock signal after passing through the channel is y _N (t) :

(7)

(7)

中的各采样时刻的冲击信号经过带通滤波器后的时域信号在所述内插时刻的叠 加值； It can be seen from Figure 3 that an interpolation value is inserted at the zero-crossing position (interpolation time) between the peak points of two symbols, and the interpolation value is determined by

The superposition value of the time-domain signal at the interpolation moment of the impact signal at each sampling moment in the bandpass filter;

That is, the value to be inserted at the time of interpolation is:

(8)

(8)

组合后，使得采 样信号恢复成2倍码元速率，使此采样序列满足了奈奎斯特采样定理。 The receiving device combines the interpolation values at different interpolation moments calculated by formula (8) with

After combination, the sampling signal is recovered to double the symbol rate, and the sampling sequence satisfies the Nyquist sampling theorem.

The following description is used to verify that the interpolation algorithm can have a good suppression effect on the first side lobe of the signal.

The interpolation algorithm based on prior filtering is simulated. The simulation conditions are: the system function of the band-limited channel is a raised cosine function, the shaping coefficient is 0.5, and the number of symbols is 1, 3, 9, 17, 25, and 31 respectively. FFT operation 1024 points.

Select the interpolated spectrograms with the number of correlation operation symbols being 1, 3, 9, 17, 25, and 31 respectively, and measure the signal spectrum of different numbers of interpolated symbols in the spectrogram, and sample 3 symbols When interpolating, the suppression of the first side lobe can only reach 20dB; when 31 symbols are used for interpolation, the suppression of the first side lobe can reach 60dB; with the increase of the number of interpolation symbols, the suppression of the first side lobe The inhibition of the valve increased continuously.

Therefore, when the number of correlation operation symbols selected by the prior filtering interpolation algorithm is greater than 30, the interpolation algorithm can have a good suppression effect on the first side lobe of the signal.

The following content is used to illustrate the improvement of the signal-to-noise ratio by the interpolation algorithm of prior filtering.

There is Gaussian white noise in the channel. Before interpolation, the Gaussian white noise signal ζ _n at any sampling time ( t=nT ) has an average value of a and a variance of σ ² . If the input to a linear system is a Gaussian signal, the output of the system is also a Gaussian signal.

After interpolation, the Gaussian white noise signal at the interpolation moment is:

After interpolation, the Gaussian white noise signal at the interpolation point is determined by its eigenvalues.

The mean of the interpolation moments is:

Because E[ζ _n ] =0 , the mean value of the random signal at the time of interpolation is also zero, that is, E[ζ _NI ] =0.

The variance of the interpolation time is:

Right now:

Expanding the above formula, each element in the expansion formula is divided into two types: the square of the same element and the product of different elements. The product of different elements represents the product of the noise signal at different times, because the Gaussian white noise signal is independent of each other at each time , so the mean of the product of different elements is zero.

Through the above analysis, the above formula can be expressed as:

The variance of the random signal represents the AC power of the signal. When the mean value is zero, the AC power of the noise signal at the interpolation moment is the total power of the signal. The above formula shows that the signal variance at the time of interpolation is closely related to the system function of the transmission channel.

The noise power at the interpolation time can be equivalent to the sum of the squares of the system function samples at different interpolation time, that is,

MATALAB software was used to calculate and compare the above formula.

In the case of different shaping coefficients, the difference value is performed on the QPSK modulated signal, and the signal-to-noise ratio deteriorates as shown in Table 1.

Table 1 The deterioration of noise power with different shaping coefficients for double interpolation

During the 31-symbol correlation calculation, since the interpolation value is the zero-crossing point value and is interpolated at the non-peak point of the signal, the noise value at the interpolation point is not only not deteriorated, but is better than direct sampling. The larger the shaping coefficient of the signal, the smaller the smear after filtering, the smaller the noise power of the response, and the more obvious the suppression of noise during interpolation; when the shaping coefficient is 0.1, the signal-to-noise ratio of the interpolation point is directly The sampling improvement is 0.39dB; when the shaping factor is 0.9, the signal-to-noise ratio of the interpolation point is improved by 2.6dB.

Therefore, the prior filtering interpolation algorithm can improve the signal-to-noise ratio.

Other embodiments of the invention will be readily apparent to those skilled in the art from consideration of the specification and practice of the invention disclosed herein. This application is intended to cover any modification, use or adaptation of the present invention, these modifications, uses or adaptations follow the general principles of the present invention and include common knowledge or conventional technical means in the technical field not disclosed in this disclosure . The specification and examples are to be considered exemplary only, with a true scope and spirit of the invention being indicated by the following claims.

It should be understood that the present invention is not limited to the precise constructions which have been described above and shown in the accompanying drawings, and various modifications and changes may be made without departing from the scope thereof. The scope of the invention is limited only by the appended claims.

Claims (7)

1. A single code element rate sampling method based on prior filtering applied to high-speed transmission is applied to receiver equipment, the receiver equipment is satellite equipment or ground equipment, the receiver equipment comprises an analog-to-digital conversion chip and a digital processing chip, and the method is characterized by comprising the following steps:

the analog-to-digital conversion chip processes the received time domain continuous signal to obtain a first sampling signal; wherein the first sampled signal corresponds to a first sampling frequency, the first sampling frequency being a single symbol rate;

the analog-to-digital conversion chip sends the first sampling signal to a digital processing chip;

the digital processing chip calculates an interpolation value at each interpolation time according to the first sampling signal, and combines the first sampling signal and the interpolation value into a second sampling signal according to the interpolation time; wherein the second sampled signal corresponds to a second sampling frequency, the second sampling frequency being 2 times a symbol rate;

the sender equipment samples the time domain continuous signal by using M times of first sampling frequency to obtain a third sampling signal; wherein M is an integer greater than 1; wherein the first sampling frequency is a single symbol rate;

setting M-1 sampling points at the same position in continuous M sampling points corresponding to each code element in the third sampling signal to be zero to obtain a fourth sampling signal;

and transmitting a time domain continuous signal corresponding to the fourth sampling signal.

2. The method of claim 1,

the analog-to-digital conversion chip processes the received time domain continuous signal to obtain a first sampling signal, and the method comprises the following steps:

the analog-to-digital conversion chip samples the time domain continuous signal by using a first sampling frequency to obtain a first sampling signal; wherein the first sampling frequency is a single symbol rate.

3. The method of claim 1,

the interpolation instant is a zero-crossing position in the middle of the peak points of the respective two adjacent symbols in the first sampled signal.

4. The method of claim 1,

the interpolation value at the interpolation time is the superposition value of the time domain signals of the impact signals of all sampling times in the first sampling signals after passing through the band-pass filter at the interpolation time.

5. The method of claim 1,

the method further comprises the following steps:

testing the frequency spectrum response of a target channel by using a test signal known by both receiver equipment and sender equipment, and determining the frequency spectrum characteristic of a band-pass filter corresponding to the target channel;

wherein the test signal contains all spectral components in-band of the target channel; the target channel comprises a shaping filter of a sending device, a wireless channel used for transmitting wireless data and a matched filter of a receiving device, and the target channel is a linear time-invariant channel.

6. The method of claim 5,

the system function of the band-pass filter is a raised cosine function, the forming coefficient is 0.5, and the number of corresponding code elements is more than 30.

7. The method of claim 6,

the number of symbols is 31.

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