Amplitude distortion automatic calibration method and device, and medium

Abstract

The invention discloses an amplitude distortion automatic calibration method and device, and a medium, which are applied to an upper computer, and the method comprises the steps: obtaining an in-band dot frequency signal generated by a signal source, and converting the in-band dot frequency signal into a digital signal; digital signal power is calculated, and a digital power vector is generated; according to the order of the filter and the bandwidth parameter, determining the order of the compensation filter and a transition band range; calculating a compensation vector based on the digital power vector, generating a phase vector according to the filter order and the interpolated digital power vector point number, and calculating the compensation vector and the phase vector to obtain a compensation frequency response; performing time-frequency conversion on the compensation frequency response, and determining a compensation filter coefficient; and carrying out normalization processing on the compensation filter coefficient, and loading the coefficient to the compensation filter to complete the automatic calibration of the amplitude distortion of the receiving channel. According to the invention, the flexibility is improved, the resource consumption of a hardware multiplier is reduced, and the problem of amplitude response distortion caused by in-band fluctuation is solved.

Description

Technical Field

[0001] This invention relates to the field of digital signal processing technology, and in particular to an automatic amplitude distortion calibration method, apparatus, and medium. Background Technology

[0002] With the continuous development of modern wireless communication technology, the operating frequency band of communication equipment is becoming wider and wider. For communication equipment, the wider the operating frequency band, the more difficult it is for complex hardware such as power amplifiers and multiplexers to match it. As a result, the in-band fluctuations caused by this are aggravated by cascading and can lead to significant distortion, affecting network applications.

[0003] In existing technical solutions, the frequency sampling method obtains the system's frequency response by sweeping the system's frequency, then compensates for the system's frequency response and performs time-frequency conversion to obtain the fitting filter coefficients. The target frequency response and the system frequency response are used to calculate the compensated frequency response. The compensated frequency response is then converted to time-frequency to obtain the fitting filter coefficients, which are then imported into the signal link to compensate the system and achieve the effect of eliminating in-band ripples. In addition, the adaptive algorithm compensates for the system response by continuously iterating the adaptive filter, thereby achieving the effect of eliminating in-band ripples.

[0004] The patent "A Method and System for Improving In-Band Flatness of Network Optimization Equipment" introduces CPU to compensate for the system frequency response, increasing system overhead and failing to address the optimization of FPGA hardware multiplier resource consumption by complex filters. The patent "Ultra-Wideband Adaptive Fluctuation Compensation Method and System" uses third-party software MATLAB to generate filter coefficients, sweeps the system frequency, and collects the frequency response of the entire system to obtain in-band fluctuation data. An IFFT is then performed on this data to obtain a compensation filter for fluctuation compensation, which is not conducive to automation and mass production development. The patent "A Broadband Calibration Compensation Method for Receiver Channel" uses a frequency sampling method that performs time-frequency conversion on the data from the signal source and the data from the receiver channel to obtain the compensated amplitude-frequency response. A compensation filter is then obtained based on the compensated amplitude-frequency response to complete the compensation of the receiver channel. However, this method requires performing FFT on the signal generated by the signal source and the signal received by the receiver to obtain the amplitude compensation response and phase compensation response, and then obtaining the filter to compensate and calibrate the receiver channel, making the implementation process quite complex. Summary of the Invention

[0005] In view of the shortcomings of the prior art, the present invention provides an automatic amplitude distortion calibration method, device and medium, which improves flexibility, reduces hardware multiplier resource consumption and solves the problem of amplitude response distortion caused by in-band ripple.

[0006] To address the aforementioned technical problems, the first aspect of this invention discloses an automatic amplitude distortion calibration method, applied to a host computer, the method comprising: Acquire the in-band frequency signal generated by the signal source and convert the in-band frequency signal into a digital signal; Calculate the power of the digital signal and generate a digital power vector; Based on the filter order and bandwidth parameters, determine the order of the compensation filter and the transition band range; The compensation vector is calculated based on the digital power vector. A phase vector is generated according to the filter order and the number of points of the interpolated compensation vector. The interpolated compensation vector and the phase vector are then used to obtain the compensation frequency response. The compensation frequency response is converted to a time-frequency value to determine the compensation filter coefficients; The coefficients of the compensation filter are normalized and loaded into the compensation filter to complete the automatic calibration of amplitude distortion in the receiving channel.

[0007] In some implementations, when both the filter coefficients and the receiving channel frequency response are complex numbers, the filter coefficients and the receiving channel frequency response are calculated by decomposing complex multiplication. When the frequency response of the receiving channel is a real number, the real part of the complex compensation filter is taken.

[0008] In some implementations, when both the filter coefficients and the receiving channel frequency response are complex numbers, the filter coefficients and the receiving channel frequency response are calculated by decomposing complex multiplication, including:

[0009]

[0010] in, , Let be the real and imaginary parts of the filter coefficients. , Let N be the I and Q components of the input signal, N be the filter order, and k be the filter index.

[0011] In some implementations, calculating the compensation vector based on the digital power vector includes: The mean of the digital power vector is calculated based on the digital power vector, and the compensation vector is calculated based on the mean of the digital power vector.

[0012] In some implementations, the compensation vector is calculated using the following formula: ; in, Let A be the mean of the digital power vectors.

[0013] The phase vector is generated by the following formula;

[0014] Where p1 is the frequency index vector; N is the filter order; The compensated frequency response is calculated using the following formula:

[0015] Where A2 is the desired amplitude response and P is the phase vector.

[0016] In some implementations, the order of the compensation filter and the transition band range are determined based on the filter order and bandwidth parameters, including: Transition bands are set at the start and cutoff frequencies of the passband, and uniform interpolation or smoothing is performed. Set the stopband to zero to form the complete desired amplitude response.

[0017] In some implementations, time-frequency conversion is performed on the compensated frequency response to determine the compensation filter coefficients, including: The compensation frequency response is converted to a time-frequency signal, and the first N+1 bits of the time-frequency conversion result are extracted to generate the compensation filter coefficients; where N is the filter order.

[0018] In some implementations, the digital power vector is calculated using the following formula:

[0019] Where I and Q are the real and imaginary components of the digital signal itself.

[0020] Secondly, an automatic amplitude distortion calibration device is disclosed, including: A signal source is used to generate a point-frequency signal that is uniformly distributed within the system passband. The power calculation module is used to convert the analog-to-digital converted digital signal into a digital power vector; The host computer includes a control module, a filter coefficient calculation module, and a filter coefficient loading module; the filter coefficient calculation module is used to generate a compensation frequency response based on the digital power vector and generate compensation filter coefficients through time-frequency conversion; The filter coefficient loading module is used to normalize and gain-balance the compensation filter coefficients and then load them into the compensation filter module. The compensation filter module, based on an FIR filter, is used to calibrate the amplitude distortion of the receiving channel in real time.

[0021] Thirdly, a computer storage medium is disclosed, on which a computer program is stored, which, when executed by a processor, implements the amplitude distortion automatic calibration method as described in any of the above.

[0022] Compared with the prior art, the beneficial effects of the present invention are as follows: This invention provides an automatic amplitude distortion calibration method, device, and medium. It automatically acquires and analyzes in-band signal power via a host computer, dynamically generates compensation filter coefficients, and applies these coefficients in real-time using phase linear optimization technology, achieving efficient calibration of amplitude distortion in the receiving channel. By simplifying complex number operations to reduce hardware resource consumption, and utilizing dynamic mean compensation to improve passband flatness, coupled with fully automated control, calibration time is shortened and mass production consistency is improved. This effectively solves the signal distortion problem caused by hardware mismatch in broadband communication systems and is suitable for communication and radar equipment with high bandwidth and high stability requirements. Attached Figure Description

[0023] Figure 1 This is a flowchart illustrating the automatic amplitude distortion calibration method provided by the present invention. Figure 2 A schematic diagram of the compensation filter for the automatic amplitude distortion calibration method provided by the present invention; Figure 3 This is a schematic diagram of the automatic amplitude distortion calibration device provided by the present invention. Detailed Implementation

[0024] To better understand and implement this invention, the technical solutions in the embodiments of this invention will be clearly and completely described below with reference to the accompanying drawings. Obviously, the described embodiments are only a part of the embodiments of this invention, and not all of them. All other embodiments obtained by those skilled in the art based on the embodiments of this invention without creative effort are within the scope of protection of this invention.

[0025] The terms “comprising” and “having” and any variations thereof in this invention are intended to cover non-exclusive inclusion, for example, a process, method, system, product or device that includes a series of steps or modules is not necessarily limited to those steps or modules that are explicitly listed, but may include other steps or modules that are not explicitly listed or that are inherent to such processes, methods, products or devices.

[0026] This invention discloses an automatic amplitude distortion calibration method, which implements calibration through a host computer, improving flexibility, reducing hardware multiplier resource consumption, and only increasing the time delay overhead of the compensation filter for the signal link. It also solves the problem of amplitude response distortion caused by in-band ripple, achieving a better amplitude distortion calibration effect.

[0027] like Figure 1 As shown, this method is executed by the host computer and includes: Step S1: Obtain the in-band frequency signal generated by the signal source and convert the in-band frequency signal into a digital signal; The signal source is connected to the near and far ends, generating a frequency with uniform intervals. The signal is an in-band frequency signal, for example, 10 frequency signals are generated at 10MHz intervals within the 0~100MHz frequency band. At the same time, the power amplifier module amplifies the in-band frequency signal and outputs it to the receiving channel to simulate the amplitude distortion in the actual communication link. The analog signal is converted into a digital signal by the analog-to-digital converter module, and the compensation filter module is set to bypass mode.

[0028] Step S2: Calculate the digital signal power and generate a digital power vector.

[0029] The digital signal power is calculated based on the digital signal converted by the analog-to-digital converter module and then transmitted to the control module. Specifically, the digital signal power is calculated using the following formula:

[0030] Where I and Q are the real and imaginary components of the digital signal itself.

[0031] The control module reads the digital power vector corresponding to the signal with internal frequency. The digital power vector is a vector composed of digital power obtained from inputting point-frequency signals at different frequencies within the band to the power calculation module. After obtaining the digital power vector, the control module sends it to the filter coefficient calculation module and displays it on the GUI interface. The GUI interface is used to display parameters such as filter coefficients and filter amplitude. Simultaneously, users can view the data they need or input relevant filter parameters through the GUI interface to facilitate subsequent automatic amplitude calibration.

[0032] Step S3: Determine the order of the compensation filter and the transition band range based on the filter order and bandwidth parameters.

[0033] The filter order and bandwidth parameters can be transmitted to the control module via a GUI interface, and can be determined based on specific hardware requirements. The GUI interface provides editable input windows for core parameters such as filter order, bandwidth, and sampling interval, allowing users to directly input and adjust parameter values. After verification, the input parameters are transmitted to the control module in real time via serial port / UDP protocol, ensuring immediate response during the calibration process. The GUI interface enables rapid configuration of filter parameters, avoiding the complexity of traditional command-line or script operations, improving the human-computer interaction efficiency of the calibration process, and reducing the risk of misoperation. The filter order is selected based on actual conditions, such as dynamically choosing the minimum order based on bandwidth parameters and stopband suppression requirements.

[0034] The smooth transition region, or transition band range, is automatically generated based on the filter order N. Specifically, it can include: Step S31: Set transition bands at the start and cutoff frequencies of the passband, and perform uniform interpolation or smoothing. Step S32: Set the stopband portion to zero to form the complete desired amplitude response.

[0035] When the sampling frequency is Fs, the bandwidth of the filter transition band is determined according to actual requirements. The two transition zones are respectively , , and The signal can be drawn as a uniformly spaced straight line or with appropriate smoothing. Setting the stopband to an appropriate number of zeros yields the desired amplitude response. The amplitude is forced to zero to avoid signal leakage.

[0036] The transition band range can be automatically generated based on the filter order N. For example, when the filter order N is 64, the transition band bandwidth can be 0.1. The passband is located on both sides, such as 80~90MHz and 110~120MHz. A stopband is set outside the transition band, such as <80MHz and >120MHz, with the amplitude forced to zero to avoid signal leakage.

[0037] Step S4: Calculate the compensation vector based on the digital power vector, generate a phase vector according to the filter order and the number of points of the interpolated compensation vector, and perform operations on the interpolated compensation vector and the phase vector to obtain the compensation frequency response.

[0038] Calculate the mean of the digital power vector based on the digital power vector. The compensation vector is calculated based on the mean of the digital power vector to smooth out passband ripples. Specifically, the compensation vector is calculated using the following formula: ; in, Let A be the mean of the digital power vectors.

[0039] The phase vector is generated by the following formula;

[0040] Where p1 is the frequency index vector, i.e. the total number of sampling points, [-1:1], and its length is consistent with the expected amplitude response A2; N is the filter order; The compensated frequency response H is calculated using the following formula:

[0041] Where A2 is the desired amplitude response and P is the phase vector.

[0042] Step S5: Perform time-frequency conversion on the compensation frequency response to determine the compensation filter coefficients.

[0043] The compensated frequency response is subjected to time-frequency conversion, and an inverse fast Fourier transform (IFFT) is performed on H to obtain the time-frequency conversion result, which is the time-domain complex coefficients. The first N+1 bits of the time-frequency conversion result are extracted to generate the compensation filter coefficients, and the tail is truncated to reduce resource consumption.

[0044] In this application, as Figure 2 As shown, resource optimization of the multiplier is performed in the implementation of the compensation filter module. The expression for the FIR filter is:

[0045] Where H(k) are the filter coefficients and X(k) are the frequency response of the receiving channel.

[0046] When the receiving channel frequency response X(k) is a real number, meaning the frequency response is symmetrical about zero frequency, compensation for amplitude distortion in the receiving channel frequency response is performed from a resource optimization perspective, and the filter coefficients H(k) are real numbers. When the receiving channel frequency response X(k) is a real number, it is assumed that the amplitude distortion of the receiving channel is symmetrical about the center frequency. In this case, the method can still be used to acquire the digital power vector, then calculate the compensation vector and phase vector to obtain the compensation frequency response. After time-frequency conversion of the compensation frequency response, complex compensation filter coefficients are obtained. Finally, the real part of the complex compensation filter coefficients is taken into the compensation filter to complete the amplitude distortion compensation of the receiving channel.

[0047] When the frequency response X(k) of the receiving channel is a complex number and the filter coefficient H(k) is a real number, the frequency response of the filter coefficient H(k) is symmetrical about zero frequency, while the frequency response of the receiving channel is asymmetrical about zero frequency. In this case, the compensation provided by the filter for the receiving channel is not comprehensive.

[0048] When both the filter coefficients H(k) and the receiving channel frequency response X(k) are complex numbers, the multiplication operation of the complex FIR filter is decomposed into real number operations: Let H(k)=a(k)+jb(k), X(k)=c(k)+jd(k):

[0049]

[0050] for

[0051] It can be sorted out.

[0052]

[0053] Substituting this into the complex FIR filter expression, we get:

[0054]

[0055]

[0056] in

[0057]

[0058]

[0059] Will , and By combining and extracting common factors and recombination terms, Y(n) is calculated, yielding:

[0060]

[0061] Each order requires 3 multipliers, and an Nth order requires a total of 3N multipliers. Compared to existing methods:

[0062] 4N multipliers are required. This method saves one multiplier per order, thus saving N multipliers and 25% of multiplier resources. N is the filter order. By reconstructing the complex multiplication logic, hardware resource consumption is reduced while maintaining performance, making it suitable for low-power, low-cost deployment in broadband communication systems.

[0063] The filter coefficient calculation module normalizes the calculated fitted filter coefficients and then transmits them to the filter coefficient loading module.

[0064] Step S6: Normalize the coefficients of the compensation filter and load them into the compensation filter module to complete the automatic calibration of amplitude distortion in the receiving channel.

[0065] The filter coefficient loading module quantizes the normalized fitted filter coefficients and levels the gain of the filter coefficients according to the mean of the digital power vector. The compensation filter module loads the fitted filter coefficients quantized by the filter coefficient loading module to complete the fluctuation compensation of the receiving channel.

[0066] This application combines filter order and bandwidth parameters to dynamically design the transition band range and generate optimized compensation coefficients, thereby reducing hardware resource consumption while ensuring calibration accuracy. The automated parameter configuration and real-time application functionality on the host computer further improves engineering deployment efficiency, making it suitable for large-scale mass production scenarios.

[0067] Compared to the original solution, the calculation process of the fitting filter coefficients in this invention is implemented by the host computer, which improves flexibility, reduces the consumption of hardware multiplier resources, and only increases the time delay overhead of the compensation filter for the signal link. It also solves the problem of amplitude response distortion caused by in-band ripple, and achieves better amplitude distortion calibration effect.

[0068] Based on the same inventive idea, such as Figure 3 As shown, this application also provides an automatic amplitude distortion calibration device, comprising: A signal source is used to generate a point-frequency signal that is uniformly distributed within the system passband. The power calculation module is used to convert the analog-to-digital converted digital signal into a digital power vector; The host computer includes a control module, a filter coefficient calculation module, and a filter coefficient loading module; the filter coefficient calculation module is used to generate a compensation frequency response based on the digital power vector and generate compensation filter coefficients through time-frequency conversion; The filter coefficient loading module is used to normalize and gain-balance the compensation filter coefficients and then load them into the compensation filter module. The compensation filter module, based on an FIR filter, is used to calibrate the amplitude distortion of the receiving channel in real time.

[0069] The control module controls the signal source to generate uniformly spaced in-band frequency signals. The power calculation module calculates the digital power of the digital signal converted by the analog-to-digital converter. After reading the digital power vector corresponding to the in-band frequency, the control module sends the digital power vector to the filter coefficient calculation module and displays it on the GUI interface. To maintain filter linearity, the filter coefficient calculation module calculates the compensation vector from the obtained digital power vector and then calculates the corresponding phase vector based on the frequency of the digital power vector and the filter order. The compensation frequency response is obtained from the compensation vector and phase vector. This response is then converted to a time-frequency signal, and the first N+1 points of the IFFT result are extracted to obtain the compensation filter coefficients. The filter coefficient loading module imports the coefficients of the compensation filter module, thus achieving amplitude distortion compensation for the receiving channel.

[0070] Based on the same inventive concept, the present invention also provides a computer device, comprising: a processor and a memory; wherein the memory stores a computer program adapted to be loaded by the processor and to execute the steps of the above-described automatic amplitude distortion calibration method.

[0071] The processing methods for computer devices can be referred to the description of the methods above, and will not be repeated here.

[0072] This application also provides a non-transitory machine-readable storage medium storing an executable program, which, when run by a microprocessor, causes the processor to execute the method provided in the above embodiments.

[0073] This invention discloses a computer-readable storage medium storing a computer program for electronic data interchange, wherein the computer program causes a computer to perform the described methods.

[0074] This invention discloses a computer program product including a non-transitory computer-readable storage medium storing a computer program operable to cause a computer to perform the described method.

[0075] The embodiments described above are merely illustrative. The modules described as separate components may or may not be physically separate. The components shown as modules may or may not be physical modules; that is, they may be located in one place or distributed across multiple network modules. Some or all of the modules can be selected to achieve the purpose of this embodiment according to actual needs. Those skilled in the art can understand and implement this without any creative effort.

[0076] Through the detailed description of the above embodiments, those skilled in the art can clearly understand that each implementation method can be implemented by means of software plus necessary general-purpose hardware platforms, and of course, it can also be implemented by hardware. Based on this understanding, the above technical solutions, in essence or the part that contributes to the prior art, can be embodied in the form of a software product. This computer software product can be stored in a computer-readable storage medium, including read-only memory (ROM), random access memory (RAM), programmable read-only memory (PROM), erasable programmable read-only memory (EPROM), one-time programmable read-only memory (OTPROM), electrically-Erasable Programmable Read-Only Memory (EEPROM), compact disc read-only memory (CD-ROM) or other optical disc storage, disk storage, magnetic tape storage, or any other computer-readable medium that can be used to carry or store data.

[0077] Finally, it should be noted that the embodiments disclosed in this invention are merely preferred embodiments of this invention and are only used to illustrate the technical solutions of this invention, not to limit it. Although this invention has been described in detail with reference to the foregoing embodiments, those skilled in the art should understand that modifications can still be made to the technical solutions described in the foregoing embodiments, or equivalent substitutions can be made to some of the technical features. Such modifications or substitutions do not cause the essence of the corresponding technical solutions to deviate from the spirit and scope of the technical solutions of the embodiments of this invention.

Claims

1. A method of automatic amplitude distortion calibration, characterized in that, The method is applied to an upper computer, and the method comprises the following steps: An in-band point frequency signal generated by a signal source is acquired, and the in-band point frequency signal is converted into a digital signal; The power of the digital signal is calculated to generate a digital power vector; The order of a compensation filter and a transition band range are determined according to a filter order and a bandwidth parameter; A compensation vector is calculated based on the digital power vector, a phase vector is generated according to the filter order and the point number of the interpolated compensation vector, the interpolated compensation vector is operated with the phase vector, and a compensation frequency response is obtained; The compensation frequency response is subjected to time-frequency conversion to determine compensation filter coefficients; The compensation filter coefficients are normalized and loaded into the compensation filter to complete automatic calibration of amplitude distortion of a receiving channel.

2. The amplitude distortion automatic calibration method of claim 1, wherein, When the filter coefficients and the receiving channel frequency response are complex numbers, the compensation filter is calculated through decomposition of complex number multiplication; When the receiving channel frequency response is a real number, the real part of the compensation filter is taken.

3. The amplitude distortion auto-calibration method of claim 2, wherein, When the filter coefficients and the receiving channel frequency response are complex numbers, the filter coefficients and the receiving channel frequency response are calculated through decomposition of complex number multiplication, which comprises the following steps: wherein , are real and imaginary parts of filter coefficients, , are I, Q components of an input signal, N is a filter order, and k is a filter index.

4. The amplitude distortion auto-calibration method of claim 2, wherein, The compensation vector is calculated based on the digital power vector, which comprises the following steps: The digital power vector mean value is calculated based on the digital power vector, and the compensation vector is calculated according to the digital power vector mean value.

5. The amplitude distortion auto-calibration method of claim 4, wherein, The compensation vector is calculated through the following formula: ； wherein is the digital power vector mean, A is the digital power vector; The phase vector is generated through the following formula: Wherein, p1 is a frequency index vector; N is the filter order; The compensation frequency response is calculated through the following formula: Wherein, A2 is the expected amplitude response, and P is the phase vector.

6. The amplitude distortion auto-calibration method of claim 3, wherein, The order of the compensation filter and the transition band range are determined according to the filter order and the bandwidth parameter, which comprises the following steps: The transition bands are respectively set at the passband start frequency and the cutoff frequency, and uniform interpolation or smoothing processing is performed; The stopband part is set to zero to form a complete expected amplitude response.

7. The amplitude distortion automatic calibration method of claim 3, wherein, The compensation frequency response is subjected to time-frequency conversion to determine the compensation filter coefficients, which comprises the following steps: The compensation filter coefficients are generated by performing time-frequency conversion on the compensation frequency response and intercepting the first N+1 bits of the time-frequency conversion result; wherein N is the filter order.

8. The amplitude distortion auto-calibration method of claim 6, wherein, The digital power vector is calculated through the following formula: Wherein, I and Q are the real part and the imaginary part of the digital signal itself.

9. Amplitude distortion auto-calibration apparatus, characterized in that It comprises the following steps: A signal source is used to generate point frequency signals uniformly distributed in a system passband; A power calculation module is used to convert the digital signal after analog-digital conversion into a digital power vector; The upper computer comprises a control module, a filter coefficient calculation module and a filter coefficient loading module; The filter coefficient calculation module is used to generate a compensation frequency response according to the digital power vector, and to generate compensation filter coefficients through time-frequency conversion; The filter coefficient loading module is used to load the compensation filter coefficients into the compensation filter module after normalization and gain leveling; The compensation filter module is based on a FIR filter and is used to perform real-time calibration on the amplitude distortion of a receiving channel.

10. A computer storage medium, characterized in that, A computer program is stored thereon, and the computer program is executed by a processor to realize the steps of the amplitude distortion automatic calibration method according to any one of claims 1-8.

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