Amplitude and phase calibration device

Abstract

The application discloses an amplitude and phase calibration device, which is a receiver device or a transmitter device, and comprises an acquisition module, a low-frequency calibration module and a restoration module. The acquisition module is used for acquiring source data output by a source data output end and obtaining sampling data based on the source data. The low-frequency calibration module is used for filtering the sampling data according to a filter coefficient set to obtain a difference signal. The restoration module is used for outputting compensation data matched with a type of a signal to be compensated according to the difference signal. The compensation module is used for compensating the signal to be compensated based on the compensation data to generate a target signal after amplitude and phase calibration. By using the device, high-precision low-frequency amplitude and phase calibration can be realized, so that the low-frequency amplitude and phase flatness of a system is improved, and under the same calibration precision requirement, resource consumption and demand for resources are reduced, flexibility is high, and applicability is high.

Description

Technical Field

[0001] This application relates to the field of signal processing, and more particularly to an amplitude and phase calibration apparatus. Background Technology

[0002] In signal processing, the amplitude and phase information of a signal are core bases for accurately characterizing the measured physical properties. When the system performance indicators of an instrument are highly sensitive to amplitude and phase characteristics, precise calibration of the instrument's amplitude and phase response is necessary to ensure the fidelity of signal measurement and the metrological traceability of signal generation. In wide-bandwidth systems, the amplitude and phase response in the low-frequency range often requires higher precision calibration. Especially in scenarios with high low-frequency energy or stringent requirements for system reproduction accuracy, deviations in amplitude and phase flatness can significantly impact system performance.

[0003] Existing technologies typically employ full-bandwidth high-precision amplitude and phase calibration or frequency-band calibration that separates high and low frequency bands to address the aforementioned issues. However, directly performing full-bandwidth high-precision amplitude and phase calibration drastically increases the required filter order, leading to a surge in hardware resources and increased implementation costs. On the other hand, with frequency-band calibration, when the low-frequency signal bandwidth is sufficiently low, it is difficult for both hardware and filters to separate the high and low frequency bands. Achieving an ideal roll-off in the low-frequency band is difficult to achieve in hardware, and achieving a narrow transition band and large roll-off rate in the filter also significantly increases the filter order, resulting in low engineering feasibility and limited applicability. Therefore, balancing hardware resources with high-precision low-frequency response calibration is a pressing issue that needs to be addressed in the amplitude and phase calibration of instruments. Summary of the Invention

[0004] This application discloses an amplitude and phase calibration device that can achieve high-precision low-frequency amplitude and phase calibration to improve the low-frequency amplitude and phase flatness of the system. At the same time, it reduces resource consumption and resource requirements under the same calibration accuracy requirements, and is highly flexible and applicable.

[0005] In a first aspect, this application provides an amplitude and phase calibration apparatus, which is a receiver apparatus or a transmitter apparatus; the receiver apparatus includes an analog front-end module, an analog-to-digital conversion module, a plurality of first calibration modules, a first compensation module, and one or more digital signal processing modules. The analog front-end module is used to output a first start signal to the analog-to-digital conversion module. The analog-to-digital conversion module is used to perform analog-to-digital conversion on the first start signal to obtain a digital calibration signal and output it to the plurality of first calibration modules. The plurality of first calibration modules are used to perform calibration processing on the digital calibration signal and output a calibrated first processed signal to the one or more digital signal processing modules. The signal processing module is used to process the first processed signal. The transmitter device includes a signal generation module, multiple second calibration modules, a second compensation module, a digital-to-analog converter module, and one or more analog signal processing circuits. The signal generation module outputs a second starting signal to the multiple second calibration modules. The multiple second calibration modules perform calibration processing on the second starting signal and output a second processed signal to the digital-to-analog converter module. The digital-to-analog converter module performs digital-to-analog conversion on the second processed signal to obtain an analog calibration signal and outputs it to the one or more analog signal processing circuits. The one or more analog signal processing circuits perform signal processing on the analog calibration signal. The aforementioned device further includes an acquisition module, a low-frequency calibration module, and a restoration module. The acquisition module acquires source data output from the source data output terminal and obtains sampled data based on the source data. The first sampling rate of the sampled data is less than the second sampling rate of the source data. The low-frequency calibration module filters the sampled data according to a set of filter coefficients to obtain a differential signal. The set of filter coefficients includes multiple filter coefficients. The set of filter coefficients reflects the difference between the target amplitude-phase flatness and the amplitude-phase flatness to be compensated in the system. The differential signal is the difference in signal quantity between the signal to be compensated under the target amplitude-phase flatness and the signal under the amplitude-phase flatness to be compensated. When the aforementioned device is a receiving... When the device is a transmitter, the signal to be compensated is any signal between the input terminal of the analog front-end module and the output terminal of the one or more digital signal processing modules; when the device is a transmitter, the signal to be compensated is any signal between the input terminal of the signal generation module and the output terminal of the one or more analog signal processing circuits; the restoration module is used to output compensation data matching the type of the signal to be compensated according to the difference signal, and the compensation data is used to merge it onto the signal to be compensated to achieve amplitude and phase calibration; both the first compensation module and the second compensation module are used to compensate the signal to be compensated based on the compensation data to generate a target signal after amplitude and phase calibration.

[0006] In this application, the receiver device may include an analog front-end module, an analog-to-digital conversion module, multiple first calibration modules, a first compensation module, and one or more digital signal processing modules. The transmitter device may include a signal generation module, multiple second calibration modules, a second compensation module, a digital-to-analog conversion module, and one or more analog signal processing circuits. Both the receiver device and the transmitter device include an acquisition module, a low-frequency calibration module, and a restoration module. The acquisition module can acquire source data output from the source data output terminal and acquire sampled data with a first sampling rate based on the source data. Here, the first sampling rate is lower than the second sampling rate of the source data. The low-frequency calibration module can filter the sampled data according to a set of filter coefficients to obtain a difference signal. Here, the difference signal can be the difference in signal quantity between the signal to be compensated under the target amplitude-phase flatness and the signal under the amplitude-phase flatness to be compensated. In the receiver device, the signal to be compensated can be any signal between the input terminal of the analog front-end module and the output terminal of the one or more digital signal processing modules. In the transmitter device, the signal to be compensated can be any signal between the input terminal of the aforementioned signal generation module and the output terminal of the aforementioned one or more analog signal processing circuits, allowing for high flexibility in the selection of the signal to be compensated. Since calibration accuracy can generally be considered as the ratio of sampling rate to filter order, the acquisition module implements data downsampling. When the differential signal is obtained through filtering in the low-frequency calibration module, the sampling rate of the downsampled data is lower, allowing for higher calibration accuracy without changing the filter order. Simultaneously, because the sampling rate of the downsampled data is lower, a smaller filter order is required to achieve the same calibration accuracy, reducing resource consumption and requirements. This approach simultaneously addresses resource constraints and achieves high accuracy in low-frequency amplitude and phase calibration. The aforementioned restoration module can output compensation data that matches the type of the signal to be compensated based on the differential signal. Here, the compensation data can be merged into the signal to be compensated to achieve amplitude and phase calibration. Using the device provided in this application, high-precision low-frequency amplitude and phase calibration can be achieved to improve the low-frequency amplitude and phase flatness of the system. The device has a simple structure and high flexibility in the connection between its modules, making it highly applicable.

[0007] In one possible implementation, the filter coefficient set comprises multiple coefficients obtained based on a first reference frequency response, a second reference frequency response, and a target frequency response. The first reference frequency response is the frequency response of the original system from the starting point to the source data output point, the second reference frequency response is the frequency response of the original system from the starting point to the ending point, and the target frequency response is the desired frequency response of the original system from the starting point to the ending point. The original system is a system without compensation by the compensation module. When the device is a receiver, the compensation module is the first compensation module, the starting point is any signal terminal from the input terminal of the analog front-end module to the output terminal of the analog-to-digital converter module, and the ending point is any signal terminal after the starting point except for the first compensation module. When the device is a transmitter, the compensation module is the second compensation module, the ending point is any signal terminal from the input terminal of the digital-to-analog converter module to the output terminal of one or more analog signal processing circuits, and the starting point is any signal terminal before the ending point except for the second compensation module. In this application, the filter coefficient set used in the low-frequency calibration module can be obtained based on a first reference frequency response, a second reference frequency response, and a target frequency response. The first reference frequency response can be the frequency response of the original system from the starting point to the aforementioned source data output point, the second reference frequency response is the frequency response of the original system from the aforementioned starting point to the aforementioned ending point, and the target frequency response is the desired frequency response of the original system from the aforementioned starting point to the aforementioned ending point. For different devices, the aforementioned starting point and the aforementioned ending point can be flexibly selected, specifically determined according to the actual application scenario and the module connection method within the device, thus improving the adaptability of the device. By determining the reference frequency response through the system of various modules in the device, and then calculating the filter coefficient set, the filter coefficient set can be made more adaptable to the actual working state of each module in the device. This allows the filter coefficient set to more accurately reflect the difference between the target amplitude and phase flatness and the amplitude and phase flatness to be compensated. Furthermore, the generated difference signal can be the signal quantity difference of the signal to be compensated under the aforementioned target amplitude and phase flatness and the aforementioned amplitude and phase flatness to be compensated. Thus, the compensation data generated based on the difference signal can be used to compensate the signal to be compensated, thereby compensating for the aforementioned signal quantity difference and achieving low-frequency amplitude and phase calibration.

[0008] In one possible implementation, the aforementioned filter coefficient set comprises multiple coefficients obtained based on the difference between the target frequency response and the second reference frequency response, as well as the first reference frequency response. In this application, the filter coefficient set can be obtained based on the difference between the target frequency response and the second reference frequency response, and the first reference frequency response. This allows the filter coefficient set to more accurately reflect the difference between the target amplitude-phase flatness and the amplitude-phase flatness to be compensated. Consequently, the difference signal obtained by filtering based on the filter coefficient set represents the signal quantity difference between the signal to be compensated under the target amplitude-phase flatness and the signal quantity difference under the amplitude-phase flatness to be compensated. This allows the compensation data generated based on the difference signal to be applied to the signal to be compensated, compensating for the signal quantity difference to achieve low-frequency amplitude-phase calibration.

[0009] In some possible implementations, each filter coefficient in the aforementioned filter coefficient set is a time-domain filter coefficient or a frequency-domain filter coefficient. When the low-frequency calibration module filters the sampled data according to the filter coefficient set to obtain the difference signal, it specifically performs time-domain filtering or frequency-domain filtering on the sampled data according to the filter coefficient set to obtain the difference signal. In this application, the low-frequency calibration module can perform time-domain filtering or frequency-domain filtering on the sampled data based on the filter coefficient set to obtain the difference signal. This allows for flexible selection of the filter coefficient set and filtering tools to implement the filtering process, improving the flexibility and applicability of the device.

[0010] In one possible implementation, the first sampling rate is less than or equal to half of the second sampling rate. In this application, the first sampling rate can be limited to being less than or equal to half of the second sampling rate. Since calibration accuracy can generally be considered as the ratio of the sampling rate to the filter order plus one, the data downsampling is achieved through the acquisition module. When the differential signal is obtained through filtering in the low-frequency calibration module, the sampling rate of the downsampled data is lower, allowing for higher calibration accuracy without changing the filter order. Simultaneously, because the sampling rate of the downsampled data is lower, a smaller filter order is required to achieve the same calibration accuracy, reducing resource consumption and requirements. This simultaneously addresses resource constraints and achieves high accuracy in low-frequency amplitude and phase calibration. The acquisition module can flexibly set the value of the first sampling rate based on the operating status of the device or its modules, resulting in high flexibility in data acquisition, adaptability to changes in the device and resources, and high applicability.

[0011] In one possible implementation, the aforementioned low-frequency band is less than or equal to half the system bandwidth of the device. In this application, the low-frequency band can be defined as less than or equal to half the system bandwidth. By reducing the sampling rate, the acquisition module can shrink the processing bandwidth of the source data, thereby reducing the design order of the filter and the real-time computation load in the subsequent low-frequency calibration module. This satisfies the requirement to reduce resource consumption and resource demands within the device while maintaining the same calibration accuracy. Simultaneously, the acquisition module can flexibly set the value range of the low-frequency band based on the operating status of the device or its various modules, resulting in high flexibility in sampling data acquisition, adaptability to changes in the device and resources, and high applicability.

[0012] In one possible implementation, when the device is a receiver, the source data output terminal is any signal terminal in the main link other than the first compensation module. The main link comprises the analog front-end module, the analog-to-digital conversion module, the plurality of first calibration modules, the first compensation module, and one or more digital signal processing modules. In this application, when the device is a receiver, any signal terminal between the input terminal of the analog front-end module and the output terminal of the one or more digital signal processing modules in the main link, excluding the first compensation module, can serve as the source data output terminal to output source data. Here, the signal terminal can be the input terminal of the module, the output terminal of the module, or any signal location within the module. Source data can be output from any signal terminal between the input terminal of the analog front-end module and the output terminal of the one or more digital signal processing modules, excluding the first compensation module. This enriches the sources of source data acquisition in the receiver, allowing the acquisition module to flexibly process source data from different locations, thus improving applicability.

[0013] In one possible implementation, when the device is a transmitter, the source data output terminal is any signal terminal in the main link except for the second compensation module. The main link comprises the signal generation module, the plurality of second calibration modules, the second compensation module, the digital-to-analog conversion module, and one or more analog signal processing circuits. In this application, when the device is a transmitter, any signal terminal between the input terminal of the signal generation module in the main link and the output terminal of the one or more analog signal processing circuits, excluding the second compensation module, can serve as the source data output terminal to output source data. Here, the signal terminal can be the input terminal of the module, the output terminal of the module, or any signal location within the module. Source data can be output from any signal terminal between the input terminal of the signal generation module and the output terminal of one or more analog signal processing circuits, excluding the second compensation module. This enriches the sources of source data acquisition in the transmitter device, allowing the acquisition module to flexibly process source data from different locations, thus providing high applicability.

[0014] In one possible implementation, the source data is a digital signal, and the acquisition module is used to: filter and downsample the source data to reduce the sampling rate of the source data from a second sampling rate to a first sampling rate, thereby obtaining sampled data with the first sampling rate. In this application, the source data can be a digital signal, and the acquisition module can filter and downsample the source data to reduce the sampling rate of the source data from a second sampling rate to a first sampling rate, thereby obtaining sampled data with the first sampling rate. The use of filtering and downsampling in the acquisition module for sampled data acquisition effectively avoids introducing distortion or loss. Since calibration accuracy can usually be considered as the ratio of the sampling rate to the filter order plus one, the acquisition module achieves data downsampling. When obtaining the difference signal through filtering in the low-frequency calibration module, the sampling rate of the downsampled data is lower, allowing for higher calibration accuracy without changing the filter order. Simultaneously, because the sampling rate of the downsampled data is lower, a smaller filter order is required to achieve the same calibration accuracy, reducing resource consumption and requirements. This approach simultaneously addresses resource constraints and achieves high accuracy in low-frequency amplitude and phase calibration.

[0015] In one possible implementation, when the signal to be compensated is a digital signal, the restoration module is configured to: perform direct interpolation processing on the difference signal or perform filtered interpolation processing on the difference signal to convert the sampling rate of the difference signal to a third sampling rate of the signal to be compensated to generate first interpolated data, wherein the low-frequency portion of the first interpolated data is consistent with the low-frequency portion of the difference signal, and the energy of other frequency portions in the first interpolated data excluding the low-frequency portion is suppressed; and output the first interpolated data as compensation data. In this application, when both the source data and the compensation signal are digital signals, the restoration module can be configured to perform direct interpolation processing or filtered interpolation processing on the difference signal to convert the sampling rate of the difference signal to a third sampling rate of the signal to be compensated to generate first interpolated data, which can then be output as compensation data. Here, the low-frequency component in the first interpolated data is consistent with the low-frequency component in the differential signal, and the energy of other frequency components in the first interpolated data, excluding the low-frequency component, is suppressed. This avoids introducing interference to the subsequent low-frequency amplitude and phase compensation of the signal to be compensated, thus improving the accuracy of amplitude and phase calibration. Through interpolation processing, the sampling rate of the compensation data can be converted to the third sampling rate of the signal to be compensated, allowing the compensation data to be directly merged into the signal to be compensated with the same sampling rate. This compensates for the signal quantity difference between the signal to be compensated under the target amplitude and phase flatness and under the amplitude and phase flatness to be compensated, thereby achieving low-frequency amplitude and phase calibration.

[0016] In one possible implementation, when the signal to be compensated is an analog signal, the restoration module is configured to: perform digital-to-analog conversion on the difference signal to obtain a first converted signal; perform synchronous calibration on the first converted signal; and output the synchronously calibrated signal as compensation data; or perform direct interpolation processing on the difference signal, or perform filtered interpolation processing on the difference signal to increase the sampling rate of the difference signal to generate second interpolated data, wherein the low-frequency portion of the second interpolated data is consistent with the low-frequency portion of the difference signal, and the energy of other frequency portions in the second interpolated data excluding the low-frequency portion is suppressed; perform digital-to-analog conversion on the second interpolated data to obtain a second converted signal; perform synchronous calibration on the second converted signal; and output the synchronously calibrated signal as compensation data. In this application, when the source data is a digital signal and the signal to be compensated is an analog signal, the restoration module can directly perform digital-to-analog conversion on the difference signal to generate a first converted signal. The first converted signal, after synchronization processing, is then output as compensation data. Alternatively, the sampling rate of the difference signal can be increased to the third sampling rate of the signal to be compensated to obtain second interpolated data. This second interpolated data is then subjected to digital-to-analog conversion to generate a second converted signal. The second converted signal, after synchronization calibration, is then output as compensation data. It is understood that the generated compensation data is in analog signal form. Synchronization calibration can compensate for the link delay between the different signal links containing the signal to be compensated and the compensation data, ensuring that the compensation data and the signal to be compensated are synchronized in time. The compensation data can be directly merged with the signal to be compensated in the first / second compensation module, simplifying the merging process of the signal to be compensated and the compensation data in the subsequent first / second compensation modules. This avoids the need to introduce digital-to-analog conversion or synchronization processing in the first / second compensation modules. The first / second compensation modules can directly combine the two analog signals through combiners to generate the target signal, reducing the complexity of the first / second compensation modules. At the same time, it enriches the way compensation data is generated, and can flexibly generate compensation data according to the application scenario and resource status, making it highly applicable.

[0017] In one possible implementation, the source data is an analog signal, and the acquisition module is configured to: perform low-pass filtering and analog-to-digital conversion on the source data using a low-pass filter and an analog-to-digital converter to obtain a digital signal corresponding to the source data; filter and downsample the digital signal corresponding to the source data to reduce the sampling rate of the digital signal corresponding to the source data from a second sampling rate to a first sampling rate, thereby obtaining sampled data with the first sampling rate; or perform low-pass filtering and analog-to-digital conversion on the source data using a low-pass filter and an analog-to-digital converter with an output digital signal sampling rate of the first sampling rate to obtain a digital signal corresponding to the source data as sampled data, wherein the sampling rate of the sampled data is the first sampling rate. In this application, the source data can be an analog signal. The acquisition module can introduce a low-pass filter and an analog-to-digital converter (ADC) to perform low-pass filtering and ADC on the source data to obtain the corresponding digital signal. Then, the digital signal is filtered and downsampled to generate sampled data with a first sampling rate. Alternatively, the source data can be directly generated by performing low-pass filtering and ADC on the source data using a low-pass filter and an ADC whose output digital signal sampling rate is the first sampling rate. Here, low-pass filtering of the source data first avoids aliasing distortion caused by the ADC. If ADC is performed first after low-pass processing, and then sampled data is acquired through filtering and downsampling, distortion or loss can be effectively avoided. Furthermore, the requirements for the ADC are lower, and more ADCs can be selected, resulting in higher applicability. If sampled data is directly generated through a low-sampling-rate ADC after low-pass processing, the reconstruction module is simpler, reducing module complexity and enriching the methods for acquiring sampled data. Since calibration accuracy can generally be considered as the ratio of the sampling rate to the filter order plus one, the sampling rate of the digital signal corresponding to the source data is reduced from the second sampling rate to the first sampling rate in the acquisition module. When the difference signal is obtained through filtering in the low-frequency calibration module, the sampling rate of the downsampled data is lower, which allows for higher calibration accuracy without changing the filter order. Simultaneously, because the sampling rate of the downsampled data is lower, a smaller filter order is required to achieve the same calibration accuracy, reducing resource consumption and requirements. This approach simultaneously addresses resource constraints and achieves high accuracy in low-frequency amplitude and phase calibration.

[0018] In one possible implementation, when the signal to be compensated is a digital signal, the restoration module is configured to: perform direct interpolation processing on the difference signal or perform filtered interpolation processing on the difference signal to convert the sampling rate of the difference signal to a third sampling rate of the signal to be compensated to generate third interpolated data, wherein the low-frequency portion of the third interpolated data is consistent with the low-frequency portion of the difference signal, and the energy of other frequency portions of the third interpolated data excluding the low-frequency portion is suppressed; perform synchronous calibration on the third interpolated data, and output the synchronously calibrated data as compensation data; or perform synchronous calibration on the difference signal to obtain a calibration signal; perform direct interpolation processing on the calibration signal or perform filtered interpolation processing on the calibration signal to convert the sampling rate of the calibration signal to a third sampling rate of the signal to be compensated to generate and output compensation data, wherein the low-frequency portion of the compensation data is consistent with the low-frequency portion of the calibration signal, and the energy of other frequency portions of the compensation data excluding the low-frequency portion is suppressed. In this application, when the source data is an analog signal and the signal to be compensated is a digital signal, the restoration module can perform direct interpolation or filtered interpolation on the difference signal to convert the sampling rate of the difference signal to the third sampling rate of the signal to be compensated, thereby generating third interpolated data. This third interpolated data can then be synchronously calibrated to output compensated data. Alternatively, the restoration module can synchronously calibrate the difference signal before performing direct interpolation or filtered interpolation to obtain compensated data. Direct interpolation or filtered interpolation can increase the sampling rate of the data, ensuring that the sampling rate of the generated compensated data matches that of the signal to be compensated, allowing for direct merging with the signal to be compensated and achieving amplitude and phase calibration. The low-frequency portion of the compensated data is consistent with the low-frequency portion of the interpolated data, and the energy of other frequency components in the third interpolated data, excluding the low-frequency portion, is suppressed, preventing interference from being introduced in subsequent compensation. Meanwhile, synchronous calibration processing can compensate for the link delay between the signal to be compensated and the compensation data located in different signal links, so that the generated compensation data can be directly superimposed on the signal to be compensated, avoiding the introduction of secondary distortion, further improving the adaptability of the device, and enabling the device to adapt to complex signal processing processes, thus having high applicability. Attached Figure Description

[0019] To more clearly illustrate the technical solutions in the embodiments of this application or the prior art, the drawings used in the description of the embodiments or the prior art will be briefly introduced below. Obviously, the drawings described below are only some embodiments of this application. For those skilled in the art, other drawings can be obtained based on these drawings without creative effort.

[0020] Figure 1aAn application architecture diagram of the amplitude and phase calibration apparatus provided in the embodiments of this application;

[0021] Figure 1b Another application architecture diagram of the amplitude and phase calibration apparatus provided in the embodiments of this application;

[0022] Figure 2 This is a schematic diagram of data interaction in a receiver device provided in an embodiment of this application;

[0023] Figure 3 This is another data interaction diagram in the receiver device provided in the embodiments of this application;

[0024] Figure 4 This is a schematic diagram of data interaction in a transmitter device provided in an embodiment of this application;

[0025] Figure 5 This is another data interaction diagram in the transmitter device provided in the embodiments of this application;

[0026] Figure 6 This is another data interaction diagram in the receiver device provided in the embodiments of this application.

[0027] Explanation of reference numerals in the attached figures:

[0028] 10 - Receiver device;

[0029] 101-Hardware circuit; 102-Field-programmable gate array; 103-Memory; 104-Display; 105-Central processing unit;

[0030] 1011 - Analog front-end module; 1012 - Analog-to-digital conversion module; 1021 - Full bandwidth amplitude and phase calibration module; 1022 - First compensation module; 1023 - Digital signal processing module; 1024 - Acquisition module; 1025 - Low frequency band calibration module; 1026 - Restoration module;

[0031] 10241 - Low-pass filter; 10242 - Analog-to-digital converter; 10243 - Filter decimation submodule; 10261 - Interpolation processing submodule; 10262 - Digital-to-analog converter; 10263 - Synchronization calibration submodule.

[0032] 20 - Transmitter device;

[0033] 201 - Signal generation module; 202 - TIS digital calibration module; 203 - Second compensation module; 204 - Digital-to-analog conversion module; 205 - Analog signal processing circuit. Detailed Implementation

[0034] The technical solutions of the embodiments of this application will be clearly and completely described below with reference to the accompanying drawings. Obviously, the described embodiments are only some embodiments of this application, and not all embodiments. Based on the embodiments of this application, all other embodiments obtained by those of ordinary skill in the art without creative effort are within the scope of protection of this application.

[0035] To facilitate understanding, the following brief explanations are provided for some of the terms:

[0036] 1. Amplitude and phase calibration: Process the actual amplitude-frequency and phase-frequency characteristics to approximate the target amplitude-phase response.

[0037] 2. Amplitude and phase flatness is used to measure the degree of deviation of the amplitude and phase frequency response of a signal processing system (or device) from the ideal transmission characteristics within the nominal bandwidth (usually the passband).

[0038] 3. Amplitude and phase flatness is used to measure the uniformity or fluctuation of the amplitude and phase frequency responses (after deducting the linear delay) of a signal processing system (or device) with frequency within the bandwidth under ideal conditions.

[0039] In signal processing, the amplitude and phase information of a signal are core bases for accurately characterizing the measured physical properties. When the system performance indicators of an instrument are highly sensitive to amplitude and phase characteristics, precise calibration of the instrument's amplitude and phase response is necessary to ensure the fidelity of signal measurement and the metrological traceability of signal generation. In wide-bandwidth systems, the amplitude and phase response in the low-frequency range often requires higher precision calibration. Especially in scenarios with high low-frequency energy or stringent requirements for system reproduction accuracy, deviations in amplitude and phase flatness can significantly impact system performance.

[0040] To solve the above problems, existing technologies require full-bandwidth amplitude and phase calibration with higher precision, which leads to a sharp increase in the required filter order, resulting in a surge in hardware resources and a rise in implementation costs. Alternatively, if existing technologies employ frequency-band calibration that separates high and low frequency bands, it becomes difficult for both hardware and filters to separate the high and low frequency bands when the low-frequency signal bandwidth is sufficiently low (e.g., less than 20MHz). Achieving an ideal roll-off in the low-frequency band is difficult to implement in hardware, and achieving a narrow transition band roll-off in the filter also significantly increases the filter order, resulting in low engineering feasibility and limited applicability. Therefore, balancing hardware resources with high-precision low-frequency response calibration is a pressing issue that needs to be addressed in the amplitude and phase calibration of instruments.

[0041] This application discloses an amplitude and phase calibration device that can achieve high-precision low-frequency amplitude and phase calibration to improve the low-frequency amplitude and phase flatness of the system. Simultaneously, it reduces resource consumption and requirements while maintaining the same calibration accuracy, offering high flexibility and applicability. The amplitude and phase calibration device disclosed in this application (hereinafter referred to as the device or amplitude and phase calibration device for ease of description) can be a receiver device or a transmitter device. Here, the receiver device can be an oscilloscope, spectrum analyzer, or other equipment, and the transmitter device can be a waveform generator, signal generator, or other equipment; the specific device can be determined according to the actual application scenario and is not limited here. For ease of description, an oscilloscope is used as an example, combined with... Figure 1a and Figure 1b The application architecture of the amplitude and phase calibration device provided in the embodiments of this application will be described.

[0042] Please see Figure 1a , Figure 1a This is an application architecture diagram of the amplitude and phase calibration apparatus provided in an embodiment of this application. Figure 1a As shown, assuming the receiver device 10 provided in this embodiment is an oscilloscope, the receiver device 10 may include hardware and software components. The hardware component may include hardware circuitry 101, a field-programmable gate array (FPGA) 102, a memory 103, and a display 104. The hardware circuitry 101 includes an analog front-end module 1011 and an analog-to-digital converter module 1012. The FPGA 102 can be used to implement functional modules including, but not limited to, multiple first calibration modules, a first compensation module 1022, a digital signal processing module 1023, an acquisition module 1024, a low-frequency calibration module 1025, and a restoration module 1026. The digital signal processing module 1023 may include multiple digital signal processing sub-modules (such as a temperature compensation module), which can be determined according to the actual application scenario and are not limited here. Figure 1a As shown, the receiver device 10 may include a main link and an auxiliary link. The main link can be the core functional link in the receiver device 10, and multiple modules in the main link can be used to implement the core functional processes of the receiver device 10, such as signal reception, signal conversion, and signal processing. Multiple modules in the auxiliary link can be used to implement high-precision amplitude and phase calibration in the low-frequency band of the system. Specifically, if... Figure 1aThe receiver device 10 shown is an oscilloscope. Its core function is to receive the measured signal and acquire the analog signal, then perform analog-to-digital conversion to generate acquired data, process the acquired data, and finally convert the data into waveforms that can be displayed on the display 104. The data is then stored in the memory 103 and output to the display 104 for waveform display. The specific configuration can be determined according to the actual application scenario and is not limited here. The main link may include the aforementioned analog front-end module 1011, the aforementioned analog-to-digital conversion module 1012, the aforementioned multiple first calibration modules, the aforementioned first compensation module 1022, and the digital signal processing module 1023. The aforementioned multiple first calibration modules may include, but are not limited to, functional modules such as the full-bandwidth amplitude and phase calibration module 1021, the synchronization calibration module, and the time-interleaved sampling (TIS) digital calibration module (not fully shown in the figure). The specific configuration can be determined according to the actual application scenario and is not limited here. For ease of description, it is assumed that the first module among the multiple first calibration modules is the full-bandwidth amplitude and phase calibration module 1021. This full-bandwidth amplitude and phase calibration module 1021 can be used to perform full-bandwidth coarse amplitude and phase calibration on the input signal, i.e., low-precision full-bandwidth amplitude and phase calibration. The auxiliary link may include the aforementioned acquisition module 1024, the aforementioned low-frequency calibration module 1025, and the aforementioned restoration module 1026. The acquisition module 1024 can be used to acquire sampled data of the source data and then output it to the low-frequency calibration module 1025. The low-frequency calibration module 1025 can generate a difference signal based on the sampled data and output it to the restoration module 1026. This difference signal can be the signal quantity difference between the signal to be compensated under the target amplitude and phase flatness and the amplitude and phase flatness to be compensated. The restoration module 1026 can output compensation data matching the type of the signal to be compensated based on the difference signal. This compensation data can be used to merge with the signal to be compensated to compensate for the aforementioned signal quantity difference, so that the system achieves the aforementioned target amplitude and phase flatness.

[0043] Optionally, the signal to be compensated can be any signal between the input terminal of the analog front-end module 1011 and the output terminal of the digital signal processing module 1023. It can be understood that any signal can include the input signal, output signal, or any signal within any module between the input terminal of the analog front-end module 1011 and the output terminal of the digital signal processing module 1023. Correspondingly, the position of the first compensation module 1022 in the main link can be flexibly defined according to the selection of the signal to be compensated. For example, if the signal to be compensated is the signal output by the analog-to-digital converter module 1012, the position of the first compensation module 1022 corresponds to the area between the analog-to-digital converter module 1012 and multiple first calibration modules (not shown in the figure). The specific location can be determined according to the actual application scenario and is not limited here. For ease of description, the following explanation will take the signal to be compensated as the signal output by multiple first calibration modules as an example. It can be understood that, Figure 1a As shown, the first compensation module 1022 is located between the multiple first calibration modules and the digital signal processing module 1023.

[0044] The acquisition module 1024 is used to acquire the source data output from the source data output terminal (not shown in the figure). Here, the source data output terminal can be any signal terminal in the main link other than the first compensation module 1022 mentioned above. That is, the source data output terminal can be any signal terminal in the main link from the input terminal of the analog front-end module 1011 to the output terminal of the digital signal processing module 1023, excluding the first compensation module 1022. It can be understood that the source data can be determined according to the source data output terminal. For example, when the output terminal of the full bandwidth amplitude and phase calibration module 1021 is the source data output terminal, the source data can be the digital signal output by the full bandwidth amplitude and phase calibration module 1021. Or, when the output terminal of the analog front-end module 1011 is the source data output terminal, the source data can be the analog signal output by the analog front-end module 1011. There are no restrictions here. For ease of understanding, taking the output terminal of the full bandwidth amplitude and phase calibration module 1021 as an example, the acquisition module 1024 can acquire the data after full bandwidth amplitude and phase calibration from the full bandwidth amplitude and phase calibration module 1021 as the source data. It can be understood that the source data is a digital signal at this time. Therefore, the acquisition module 1024 can acquire the sampled data of the source data. Here, data processing such as filtering and downsampling can be performed on the source data to reduce the sampling rate from the second sampling rate to the first sampling rate, obtaining sampled data with the first sampling rate. Here, the first sampling rate can be less than or equal to half of the second sampling rate to reduce the amount of subsequent data processing. The specific value of the first sampling rate can be determined according to the actual application scenario and is not limited here. The acquisition module 1024 can output the sampled data with the first sampling rate to the low-frequency calibration module 1025.

[0045] Furthermore, the low-frequency calibration module 1025 can acquire a set of filter coefficients, which may include multiple filter coefficients. This set of filter coefficients can be used to reflect the difference between the target amplitude-phase flatness and the amplitude-phase flatness to be compensated in the system. Here, when the position of the signal to be compensated and the end are at the same position, the target amplitude-phase flatness of the system can be used to quantify the degree to which the target frequency response of the original system from the start end to the end end approximates ideal distortion-free transmission within the passband. Here, the aforementioned start end can be the analog front-end module 1011 or the analog-to-digital conversion module 1012, and the aforementioned end end can be any module located after the start end except for the first compensation module 1022. The specific choice can be determined according to the actual application scenario and is not limited here. It can be understood that the aforementioned set of filter coefficients can be used to configure the filters or filtering tools inside the low-frequency calibration module 1025. Each filter coefficient in the set of filter coefficients can be a time-domain filter coefficient or a frequency-domain filter coefficient, which can be determined according to the actual application scenario and is not limited here. For example, assuming the low-frequency calibration module 1025 uses a finite impulse response (FIR) filter, since calibration accuracy can usually be considered as the ratio of the sampling rate to the filter order plus one, the order of the FIR filter in the low-frequency calibration module 1025 can be determined based on the first sampling rate and calibration accuracy requirements. For instance, to achieve a calibration accuracy of 100,000 Hz, a 2000th-order FIR filter can be introduced into the low-frequency calibration module 1025 to process the sampling data with a first sampling rate of 195.3125 MSa / s, outputting a differential signal. The calculated filter frequency interval is approximately 97.6 kHz, which is better than the target calibration accuracy. In this case, the filter coefficient set can be the tap coefficient sequence of the 2000th-order FIR filter. The filter coefficient set should include 2001 tap coefficients, and the specific values ​​can be determined according to the actual application scenario, without limitation here. The multiple filter coefficients in the above-mentioned filter coefficient group can be used as the coefficients of a single filter; or the low-frequency calibration module 1025 can also include multiple cascaded filters, and the multiple filter coefficients in the above-mentioned filter coefficient group can also be divided into several parts, which can be used as the coefficients of each cascaded filter respectively. This application does not limit this.

[0046] Optionally, the low-frequency calibration module 1025 can directly read the filter coefficients from the configuration file stored in the memory 103, or obtain the first reference frequency response and the second reference frequency response from the configuration file stored in the memory 103 or measured from the main link, and calculate the filter coefficients in combination with the target frequency response. Here, the first reference frequency response can be the frequency response of the original system from the starting end to the source data output end (e.g., in this embodiment, the source data output end can be the full-bandwidth amplitude and phase calibration module 1021), the second reference frequency response can be the frequency response of the original system from the starting end to the ending end, and the target frequency response is the desired frequency response of the original system from the starting end to the ending end; the original system is the system that has not been compensated by the first compensation module. In this application, a system without compensation by the first compensation module (i.e., the "original system") refers to a system that does not incorporate the first compensation module of this application for compensation (since the first compensation module operates based on modules in the auxiliary link, it also does not incorporate modules in the auxiliary link). In other words, the original system is the system in the main link. This original system does not incorporate the auxiliary link or the first compensation module for amplitude and phase calibration. At this time, the outputs of multiple first calibration modules will be directly connected to the next-level modules of multiple first calibration modules in the main link (such as...). Figure 1a The digital signal processing module 1023 is connected to the receiver. For ease of description, this application will use "original system" to replace "system without compensation by the first compensation module" or "system without compensation by the second compensation module" in the following text. The specific replacement can be determined based on the application scenario of the receiver or transmitter.

[0047] Both the first and second reference frequency responses can be measured and calculated using preset known signals such as sweep signals. For details, please refer to subsequent embodiments; further elaboration is omitted here. It is understood that the target frequency response can be frequency response data input to the receiver device 10, or frequency response data stored in the memory 103. This frequency response data is a set of data, which can be in the form of a comma-separated values ​​(CSV) table, etc., and can be determined according to the actual application scenario; no limitation is imposed here. Here, the starting end can be the input or output end of either the analog front-end module 1011 or the analog-to-digital conversion module 1012, or a signal output end within each of these modules. The ending end can be the input or output end of any module after the starting end, excluding the first compensation module 1022, or a signal output end within that module; no limitation is imposed here. It is understood that the first and second reference frequency responses can be the same, and can be determined based on the starting end, ending end, and source data output end; no limitation is imposed here. For ease of description, as... Figure 1aAs shown, the following explanation will take an analog-to-digital conversion module 1012 as the starting point, multiple first calibration modules as the ending point, and a full-bandwidth amplitude-phase calibration module 1021 as the source data output point as an example. It can be understood that the first reference frequency response is the frequency response of the first system from the analog-to-digital conversion module 1012 to the full-bandwidth amplitude-phase calibration module 1021, and the second reference frequency response is the frequency response of the second system from the analog-to-digital conversion module 1012 to the multiple first calibration modules. It can be understood that the first and second reference frequency responses can be the frequency domain characteristics of the aforementioned first and second systems, respectively, and can be determined according to the actual application scenario, without limitation here. The low-frequency calibration module 1025 can filter the sampled data output by the acquisition module 1024 according to the filtering coefficient to obtain the differential signal. Here, the low-frequency calibration module 1025 can use filtering tools such as FIR filters to filter the sampled data. The selection and parameters of the specific filtering tools can be determined according to the actual application scenario, without limitation here. It is understood that the aforementioned difference signal represents the signal quantity difference between the signal to be compensated under the target amplitude-phase flatness and the amplitude-phase flatness to be compensated. The compensation data generated based on the difference signal can be used to compensate the aforementioned signal to be compensated to compensate for the aforementioned signal quantity difference. Here, the low-frequency band can refer to a frequency band starting from 0 Hz, and the bandwidth of this frequency band can be less than or equal to 1 / 2 of the system bandwidth of the receiver device 10. The aforementioned system bandwidth can be the frequency range between the two cutoff frequency points corresponding to the signal power gain decreasing by 3 dB relative to the maximum gain value in the system frequency response curve. The specific range of the low-frequency band can be determined according to the actual application scenario. For example, assuming a bandwidth of 10 GHz, the low-frequency band can be 100 MHz or 200 MHz. This application does not impose any restrictions here.

[0048] Furthermore, the restoration module 1026 can output compensation data that matches the type of the signal to be compensated based on the difference signal. Here, the signal type can include aspects such as whether the signal is analog or digital, the signal delay, and the signal sampling rate. Superposition is only possible when the two signal types match (e.g., both are digital signals with the same delay and sampling rate). Specifically, the signal to be compensated can be either analog or digital. It is understood that since the source data can also be digital or analog, the restoration module 1026 can perform restoration processing on the difference signal according to the signal types of the source data and the signal to be compensated, to generate compensation data that matches the type of the signal to be compensated. Here, the restoration processing may include interpolation processing, digital-to-analog conversion processing, synchronous calibration processing, etc., which can be determined according to the signal types of the source data and the signal to be compensated. Specific details can be found in subsequent embodiments and will not be elaborated here. For example, such as... Figure 1aAs shown, assuming the source data is a digital signal output by the full-bandwidth amplitude-phase calibration module 1021, and the signal to be compensated is a digital signal output by multiple first calibration modules, the restoration module 1026 can interpolate the difference signal to output compensation data with the same sampling rate as the signal to be compensated. This compensation data is a digital signal. It is understood that the third sampling rate of the signal to be compensated may not be equal to the second sampling rate of the source data, or it may be equal to the second sampling rate of the source data. Specifically, it can be determined based on the influence of signal processing (such as modules for adjusting the signal sampling rate among the multiple first calibration modules) between the source data output terminal and the first compensation module 1022 on the signal sampling rate, and is not limited here. Optionally, the restoration module 1026 may include a synchronous calibration submodule, which can be used to compensate for the link delay between the main link and the auxiliary link. See subsequent embodiments for details, which will not be elaborated here. It is understood that the compensation data generated by the restoration module 1026 can be output to the first compensation module 1022 to compensate the signal to be compensated.

[0049] Furthermore, the first compensation module 1022 can combine the compensation data and the signal to be compensated using an adder or combiner. For example, an adder can be used to superimpose the compensation data and the signal to be compensated to compensate for the signal difference between the signal to be compensated under the target amplitude and phase flatness and under the amplitude and phase flatness to be compensated. The specific method can be determined according to the actual application scenario and is not limited here. The target signal generated by the first compensation module 1022 can continue to be output to subsequent modules, such as... Figure 1a As shown, the first compensation module 1022 can output the target signal to the subsequent digital signal processing module 1023, so that after processing by the digital signal processing module 1023, it can be output as a waveform through the display 104. The specific output can be determined according to the actual application scenario, and no restrictions are imposed here.

[0050] Optionally, in the receiver device 10 described above, the analog front-end module 1011 on the main link may include various analog devices, such as attenuators and amplifiers; the multiple first calibration modules, the first compensation module 1022, and the digital signal processing module 1023 may be digital domain modules (i.e., modules that process digital signals); the acquisition module 1024, the low-frequency band calibration module 1025, and the restoration module 1026 on the auxiliary link may also be digital domain modules. Here, the aforementioned digital domain modules may be composed of, for example... Figure 1a The field-programmable gate array 102 shown can also be implemented by a central processing unit (CPU). For ease of understanding, please refer to [the relevant documentation / reference]. Figure 1b , Figure 1b Another application architecture diagram of the amplitude and phase calibration apparatus provided in the embodiments of this application. (See diagram below.) Figure 1bAs shown, the receiver device 10 may include a central processing unit 105, which is responsible for executing the core instructions of the software and controlling the entire device. Here, the software portion may include, but is not limited to, the aforementioned multiple first calibration modules, first compensation module 1022, digital signal processing module 1023, and other functional modules. The specific functions of these modules can be found in the foregoing embodiments and will not be elaborated upon here. Figure 1b As shown, the acquisition module 1024, low-frequency band calibration module 1025, and restoration module 1026 included in the receiver device 10 can be implemented in hardware or in software (borne on the central processing unit 105), and no limitation is made here. It can be understood that the functions of the acquisition module 1024, low-frequency band calibration module 1025, and restoration module 1026 can all be found in [reference needed]. Figure 1a The specific implementation examples are not detailed here.

[0051] To facilitate understanding, the following will combine... Figures 2 to 6 The receiver and transmitter devices for amplitude and phase calibration provided in the embodiments of this application will be described.

[0052] Example 1

[0053] Please see Figure 2 , Figure 2 This is a schematic diagram of data interaction in a receiver device provided in an embodiment of this application. Figure 2As shown, the receiver device 10 may include a main link and an auxiliary link. The main link can be the core functional link in the receiver device 10, and may include an analog front-end module 1011, an analog-to-digital conversion module 1012, multiple first calibration modules, a first compensation module 1022, and a digital signal processing module 1023. The multiple first calibration modules may include, but are not limited to, functional modules such as a full-bandwidth amplitude and phase calibration module 1021, a synchronization calibration module (not shown in the figure), and a TIS digital calibration module (not shown in the figure), which can be determined according to the actual application scenario and are not limited here. The digital signal processing module 1023 may be one or more digital signal processing sub-modules, such as a temperature compensation sub-module, which can be determined according to the actual application scenario and are not limited here. For ease of description, the following explanation will take the first module among multiple first calibration modules, the full-bandwidth amplitude and phase calibration module 1021, as an example. This full-bandwidth amplitude and phase calibration module 1021 can be used to perform lower-precision full-bandwidth amplitude and phase calibration on the input signal. That is, compared to the aforementioned higher-precision full-bandwidth amplitude and phase calibration, the amplitude and phase frequencies of the signal are adjusted at larger frequency intervals (e.g., every 150MHz). (The corresponding high-precision calibration can be adjusted at intervals such as 10kHz or even smaller.) The analog front-end module 1011 can be used to output a first starting signal to the analog-to-digital conversion module 1012. The analog-to-digital conversion module 1012 can be used to perform analog-to-digital conversion on the first starting signal to obtain a digital calibration signal and output it to multiple first calibration modules. The multiple first calibration modules can be used to calibrate the digital calibration signal and output the calibrated first processed signal. The digital signal processing module 1023 can be used to perform signal processing on the input target signal. It can be understood that, as... Figure 2As shown, the first compensation module 1022 can be used to acquire first processed signals output by multiple first calibration modules and use the first processed signals as the signal to be compensated. The first compensation module 1022 can also be used to generate a target signal based on the signal to be compensated and compensation data, so as to output the target signal to the digital signal processing module 1023. The auxiliary link may include an acquisition module 1024, a low-frequency calibration module 1025, and a restoration module 1026. The acquisition module 1024 can be used to acquire sampled data of the source data, and then output it to the low-frequency calibration module 1025. The low-frequency calibration module 1025 can generate a difference signal based on the sampled data and output it to the restoration module 1026. The difference signal can be the signal quantity difference between the signal to be compensated under the target amplitude and phase flatness and under the compensation amplitude and phase flatness. The restoration module 1026 can output compensation data matching the type of the signal to be compensated based on the difference signal and output the compensation data to the first compensation module 1022. The compensation data can be used to merge with the signal to be compensated to compensate for the signal quantity difference, so that the system achieves the target amplitude and phase flatness. It is understood that the multiple modules in the main link can be used to realize the core functions of the receiver device 10, such as signal reception, signal conversion and signal processing, while the multiple modules in the auxiliary link can be used to realize high-precision amplitude and phase calibration in the low-frequency band of the system.

[0054] Optionally, the analog front-end module 1011 on the main link of the receiver device 10 described above may include various module devices, such as attenuators and amplifiers; the multiple first calibration modules, the first compensation module 1022, the digital signal processing module 1023, the acquisition module 1024, the low-frequency band calibration module 1025, and the restoration module 1026 may all be digital domain modules. Here, the aforementioned digital domain modules may be composed of, for example... Figure 1a The field-programmable gate array 102 shown can also be implemented by, for example, Figure 1b The implementation of the central processing unit 105 shown here will not be elaborated upon further.

[0055] Optionally, in the receiver device 10, the signal to be compensated can be any signal between the input terminal of the analog front-end module 1011 and the output terminal of the digital signal processing module 1023. It is understood that since the input signal of the first compensation module 1022 is the signal to be compensated, the position of the first compensation module 1022 can be determined based on the signal position of the signal to be compensated. It is understood that any signal can include the input signal, output signal, and any signal within any module between the input terminal of the analog front-end module 1011 and the output terminal of the digital signal processing module 1023. Correspondingly, the position of the first compensation module 1022 in the main link can be flexibly defined according to the selection of the signal to be compensated. For example... Figure 2As shown, when the first processed signal output by the multiple first calibration modules is used as the signal to be compensated, the first compensation module 1022 is connected between the multiple first calibration modules and the digital signal processing module 1023.

[0056] In some feasible implementations, the acquisition module 1024 can be used to acquire the source data output from the source data output terminal, and then obtain sampled data based on the source data. Here, the source data can be a digital signal or an analog signal, which can be determined according to the actual source data output terminal, and is not limited here. In the receiver device 10, the source data output terminal can be any signal terminal in the main link other than the first compensation module 1022, that is, the source data output terminal can be any signal terminal in the main link from the input terminal of the analog front-end module 1011 to the output terminal of the digital signal processing module 1023, excluding the first compensation module 1022. It can be understood that the source data can be the input signal, output signal, and any signal terminal within any module other than the first compensation module 1022 between the input terminal of the analog front-end module 1011 and the output terminal of the digital signal processing module 1023. For example, Figure 2 As shown, the source data output terminal can be the output terminal of the full-bandwidth amplitude and phase calibration module 1021. The acquisition module 1024 can acquire the output signal of the full-bandwidth amplitude and phase calibration module 1021 as the aforementioned source data, which is a digital signal. Please refer to [link / reference needed]. Figure 3 , Figure 3 This is another data interaction diagram in the receiver device provided in the embodiments of this application. For example... Figure 3 As shown, the source data output terminal can be the output terminal of the analog front-end module 1011, and the acquisition module 1024 can acquire the first start signal output by the analog front-end module 1011 as the source data, which is an analog signal. In this embodiment, when the device is a receiver device, any signal terminal between the input terminal of the analog front-end module in the main link and the output terminal of the one or more digital signal processing modules, excluding the first compensation module, can be used as the source data output terminal to output the source data. Here, the signal terminal can be the input terminal of the module, the output terminal of the module, or any signal position inside the module. The source data can be output from any signal terminal between the input terminal of the analog front-end module and the output terminal of one or more digital signal processing modules, excluding the first compensation module, which enriches the sources of source data acquisition in the receiver device. The acquisition module can flexibly process source data from different locations, and has high applicability.

[0057] In some feasible implementations, the low-frequency calibration module 1025 can acquire a set of filter coefficients, which may include multiple filter coefficients. This set of filter coefficients can be used to filter the sampled data to obtain a differential signal. It is understood that each filter coefficient in the filter coefficient set of this application is used to construct a filter capable of obtaining the differential signal. For example, taking FIR filtering as an example, if the filter is a time-domain filter, each filter coefficient can correspond to a tap coefficient of the FIR filter; or when the filter is a frequency-domain filter, each filter coefficient can reflect the amplitude and phase adjustment at a frequency point. Each filter coefficient can be determined by the relationship between one or more measurable frequency responses and the target frequency response. The specific determination method is not limited in this application; it is only necessary to obtain the differential signal used for compensation.

[0058] The low-frequency calibration module 1025 can be generated from storage space (such as...) Figure 1aThe filter coefficient set can be directly read from the memory 103 shown. For example, it can be read directly from a storage file containing factory data, or it can be calculated based on the first reference frequency response, the second reference frequency response, and the target frequency response. Here, the first reference frequency response can be the frequency response of the original system from the starting point to the source data output point, the second reference frequency response can be the frequency response of the original system from the starting point to the ending point, and the target frequency response can be the desired frequency response of the original system from the starting point to the ending point. In the receiver device, the starting point is any signal terminal from the input terminal of the analog front-end module 1011 to the output terminal of the analog-to-digital conversion module 1012, and the ending point is any signal terminal after the starting point except for the first compensation module 1022. It is understood that the aforementioned starting point can be any input or output terminal of any module, or any signal position within a module, from the input terminal of the aforementioned analog front-end module 1011 to the output terminal of the aforementioned analog-to-digital conversion module 1012. Similarly, the aforementioned ending point can be any input or output terminal of any module, or any signal position within a module, excluding the aforementioned first compensation module 1022, after the aforementioned starting point. Optionally, the target frequency response can be the system factory configuration data provided by the manufacturer and stored in the storage space, or it can be the expected data input to the device after delivery (which can be input by the user or the manufacturer). The specific choice depends on the actual application scenario and is not limited here. Optionally, the first reference frequency response and the second reference frequency response can be the system factory configuration data provided by the manufacturer and stored in the storage space, or they can be the data measured after delivery. The specific choice depends on the actual application scenario and is not limited here. It is understood that the aforementioned first reference frequency response can be data obtained through measurement calculation. This first reference frequency response can be obtained by applying a known preset excitation signal to the measurement channel of the original system corresponding to the first reference frequency response, and based on the relationship between its output response and the amplitude and phase characteristics of the preset excitation signal. This first reference frequency response can be data in the form of tabular data, etc., and is not limited here. It is understood that the system corresponding to the first reference frequency response is the original system from the starting point to the source data output point. For example, such as... Figure 3In the receiver device shown, the source data output terminal is the output terminal of the analog front-end module 1011. Assuming the starting terminal is selected as the input terminal of the analog front-end module 1011, the original system corresponding to the first reference frequency response should be the analog front-end module 1011. To measure the first reference frequency response, a known preset excitation signal can be input to the analog front-end module 1011 and the relationship between its output response and the amplitude and phase characteristics of the preset excitation signal can be obtained to obtain the first reference frequency response. Similarly, the acquisition or measurement method of the second reference frequency response is similar to that of the first reference frequency response. The system corresponding to the second reference frequency response is the original system from the starting terminal to the ending terminal, which will not be elaborated here. It can be understood that when the system corresponding to the first reference frequency response is the same as the system corresponding to the second reference frequency response, the first reference frequency response and the second reference frequency response can be the same. In the embodiments of this application, the filter coefficient set used in the low-frequency calibration module can be obtained based on the first reference frequency response, the second reference frequency response, and the target frequency response. The first reference frequency response can be the frequency response of the original system from the starting point to the aforementioned source data output point; the second reference frequency response is the frequency response of the original system from the aforementioned starting point to the aforementioned ending point; and the target frequency response is the desired frequency response of the original system from the aforementioned starting point to the aforementioned ending point. The starting point can be either the aforementioned analog front-end module or the aforementioned analog-to-digital conversion module, and the ending point can be any module located after the aforementioned starting point, excluding the aforementioned first compensation module. In other words, both the starting point and the ending point in the receiver device can be flexibly selected, specifically determined according to the actual application scenario and the module connection method within the device, thus improving the adaptability of the device. By determining the reference frequency response through the system of various modules in the device, and then calculating the filter coefficient set, the filter coefficient set can be made more adaptable to the actual working state of each module in the device. It can more accurately reflect the difference between the target amplitude and phase flatness and the amplitude and phase flatness to be compensated. Thus, the difference signal obtained by filtering based on the filter coefficient set is the difference in signal quantity of the signal to be compensated under the conditions of the target amplitude and phase flatness and the condition of the amplitude and phase flatness to be compensated. Therefore, the compensation data generated based on the difference signal can be used to compensate the signal to be compensated, and the above-mentioned signal quantity difference is compensated to achieve low-frequency amplitude and phase calibration.

[0059] Optionally, in some feasible implementations, assuming that when using FIR filtering, the filter coefficient set can be multiple filter coefficients obtained based on the difference between the target frequency response and the second reference frequency response, and the first reference frequency response. It is understood that the filter can be implemented in the frequency domain or the time domain; therefore, the corresponding filter coefficients can be either frequency domain or time domain filter coefficients. For example, frequency domain filter coefficients can be obtained based on the target frequency response, the first reference frequency response, and the second reference frequency response, and a frequency domain filter can be constructed based on these frequency domain filter coefficients for filtering in the frequency domain; alternatively, the frequency domain coefficients can be converted to time domain coefficients (e.g., using inverse fast Fourier transform), and a time domain filter can be constructed based on these time domain coefficients for filtering in the time domain. Specifically, for the frequency domain filter coefficients, the difference between the target frequency response and the second reference frequency response can be calculated to obtain multiple intermediate difference data for each frequency point, and then the ratio of these intermediate difference data to the first reference frequency response can be used to generate each filter coefficient in the filter coefficient set. For example, in a frequency response, there are values ​​(including amplitude and phase) corresponding to multiple frequency points. Taking one frequency point as an example, assuming the value corresponding to the frequency point in the target frequency response is a complex value 1, the value of the corresponding frequency point in the first reference frequency response is 0.8 + 0.1i, and the value of the corresponding frequency point in the second reference frequency response is also 0.8 + 0.1i, then the intermediate difference data corresponding to this frequency point is 1 - (0.8 + 0.1i) = 0.2 - 0.1i. The ratio of this intermediate difference data to the corresponding frequency point value in the first reference frequency response is 3 / 13. 2 / 13i means that the filter coefficient value corresponding to this frequency point is 3 / 13. 2 / 13i; the filter coefficients for other frequencies can also be obtained using this method. The values ​​corresponding to the above frequencies can be represented by the complex numbers mentioned above to indicate amplitude and phase information, or they can be represented by two data points (one representing amplitude and one representing phase).

[0060] To make it easier to understand, let's take amplitude as an example again. Figure 2 Further explanation is needed. Assume that when using FIR filtering, the starting point is the input of the analog front-end module 1011, and the ending point is the output of multiple first calibration modules. For example... Figure 2As shown, the source data output terminal is the output terminal of the full bandwidth amplitude and phase calibration module 1021. Therefore, the system corresponding to the first reference frequency response is the system from the analog front-end module 1011 (i.e., the starting terminal) to the full bandwidth amplitude and phase calibration module 1021 (i.e., the source data output terminal). The system corresponding to the second reference frequency response is the frequency response of the original system from the analog front-end module 1011 (i.e., the starting terminal) to the output terminals (i.e., the ending terminals) of the multiple first calibration modules. The target frequency response is the desired frequency response of the original system from the analog front-end module 1011 (i.e., the starting terminal) to the output terminals (i.e., the ending terminals) of the multiple first calibration modules. It can be understood that a frequency response includes amplitude and phase values ​​corresponding to multiple frequency points. Taking the amplitude and phase value of one frequency point as an example, assuming that the amplitude and phase value corresponding to the frequency point in the target frequency response is 1, the amplitude and phase value of the corresponding frequency point in the first reference frequency response is 0.8+0.1i, and the amplitude and phase value of the corresponding frequency point in the second reference frequency response is 0.9+0.1i, then the intermediate difference data corresponding to this frequency point is 1-(0.9+0.1i) = 0.1-0.1i. The ratio of this intermediate difference data to the corresponding frequency point value in the first reference frequency response is (0.1-0.1i) / (0.8+0.1i) = 7 / 65-(9 / 65)i. That is, if a frequency domain filter is used, the filter coefficient value corresponding to this frequency point is 7 / 65-(9 / 65)i. This filter coefficient value can be used to reflect the amplitude and phase adjustment of this frequency point in the frequency domain filter. In this embodiment, the filter coefficient set can be obtained based on the difference between the target frequency response and the second reference frequency response, as well as the first reference frequency response. This allows the filter coefficient set to more accurately reflect the difference between the target amplitude-phase flatness and the amplitude-phase flatness to be compensated. Consequently, the difference signal obtained by filtering based on the filter coefficient set is the signal quantity difference of the signal to be compensated under the target amplitude-phase flatness and the amplitude-phase flatness to be compensated. This allows the compensation data generated based on the difference signal to be compensated to compensate the signal quantity difference, thereby achieving low-frequency amplitude-phase calibration.

[0061] In some feasible implementations, the first compensation module 1022 can combine the compensation data and the signal to be compensated using an adder or combiner. For example, an adder can be used to superimpose the compensation data in digital signal form and the signal to be compensated; alternatively, a combiner can be used to combine the compensation data in analog signal form and the signal to be compensated, thereby compensating for the signal quantity difference between the signal to be compensated under the target amplitude and phase flatness and the amplitude and phase flatness to be compensated, and generating the target signal. The specific implementation can be determined according to the actual application scenario and is not limited here. For example, such as... Figure 3 As shown, the signal to be compensated is a digital signal, and the compensation data is also a digital signal. The first compensation module 1022 can use an adder to superimpose the compensation data onto the signal to be compensated in order to compensate for the above signal difference and realize low-frequency amplitude and phase calibration.

[0062] Example 2

[0063] Based on the above embodiments, please refer to Figure 4 , Figure 4 This is a schematic diagram of data interaction in a transmitter device provided in an embodiment of this application. Figure 4 As shown, the transmitter device 20 may include a main link and auxiliary links. The main link can be the core functional link in the transmitter device 20, and may include a signal generation module 201, multiple second calibration modules, a second compensation module 203, a digital-to-analog conversion module 204, and an analog signal processing circuit 205. The multiple second calibration modules may include, but are not limited to, functional modules such as a full-bandwidth amplitude and phase calibration module (not shown in the figure), a synchronization calibration module (not shown in the figure), and a TIS digital calibration module 202, which can be determined according to the actual application scenario and are not limited here. The analog signal processing circuit 205 may be composed of one or more analog signal processing sub-circuits, which can be determined according to the actual application scenario and are not limited here. For ease of description, the first module among the multiple second calibration modules, the TIS digital calibration module 202, will be used as an example for subsequent explanation. This TIS digital calibration module 202 can be used to pre-calibrate the input signal in the frequency domain and time domain. The signal generation module 201 can be used to output a second start signal to multiple second calibration modules. These multiple second calibration modules can then calibrate the second start signal and output a second processed signal. The digital-to-analog conversion module 204 can perform digital-to-analog conversion on the input signal to obtain an analog calibration signal and output the analog calibration signal. The analog signal processing circuit 205 can perform analog signal processing on the input signal. It can be understood that... Figure 4As shown, the second compensation module 203 can be used to acquire second processed signals output by multiple second calibration modules and use these second processed signals as the signal to be compensated. The second compensation module 203 can also be used to generate a target signal based on the signal to be compensated and compensation data, and output the target signal to the digital-to-analog conversion module 204. The digital-to-analog conversion module 204 can perform digital-to-analog conversion on the target signal to output an analog calibration signal to the analog signal processing circuit 205. The auxiliary link may include an acquisition module 1024, a low-frequency calibration module 1025, and a restoration module 1026. The acquisition module 1024 can acquire sampled data from the source data and output it to the low-frequency calibration module 1025. The low-frequency calibration module 1025 can generate a difference signal based on the sampled data and output it to the restoration module 1026. This difference signal can be the signal quantity difference between the signal to be compensated under the target amplitude-phase flatness and the signal under the compensated amplitude-phase flatness. The restoration module 1026 can output compensation data matching the type of the signal to be compensated based on the difference signal, and output the compensation data to the second compensation module 203. This compensation data can be used to merge with the signal to be compensated to compensate for the signal difference, so that the system achieves the target amplitude and phase flatness. Here, the acquisition module 1024, the low-frequency calibration module 1025, and the restoration module 1026 can be integrated with... Figure 2 The acquisition module 1024, low-frequency calibration module 1025, and restoration module 1026 shown are the same, and will not be described in detail here. It can be understood that the multiple modules in the main link can be used to realize the core functional processes of the transmitter device 20, such as signal generation, signal processing, and signal conversion, while the multiple modules in the auxiliary link can be used to realize high-precision amplitude and phase calibration of the system's low-frequency band.

[0064] Optionally, the acquisition module 1024, low-frequency calibration module 1025, and restoration module 1026 on the auxiliary link in the transmitter device 20 described above can also be digital domain modules. Here, the aforementioned digital domain modules can be implemented by a CPU or an FPGA, and there is no limitation on this.

[0065] Optionally, the signal to be compensated can be any signal between the input terminal of the signal generation module 201 and the output terminal of the analog signal processing circuit 205. It can be understood that since the input signal of the second compensation module 203 is the signal to be compensated, the position of the second compensation module 203 can be determined based on the signal position of the signal to be compensated. For example... Figure 4 As shown, any of the above signals can include the input signal, output signal, and any signal within any module between the input terminal of the signal generation module 201 and the output terminal of the analog signal processing circuit 205. Correspondingly, the position of the second compensation module 203 in the main link can be flexibly defined according to the selection of the signal to be compensated. For example... Figure 4As shown, when the second processed signal output by multiple second calibration modules is used as the signal to be compensated, the second compensation module 203 is connected between the multiple second calibration modules and the analog signal processing circuit 205. The second compensation module 203 can receive the second processed signal (i.e., the signal to be compensated) and generate the target signal based on the compensation data output by the restoration module 1026 in the auxiliary link. For ease of understanding, please refer to [further details omitted]. Figure 5 , Figure 5 This is another data interaction diagram in the transmitter device provided in the embodiments of this application. For example... Figure 5 As shown, in the main link of the transmitter device 20, the digital-to-analog conversion module 204 can perform digital-to-analog conversion processing on the second processing signals output by multiple second calibration modules to output the signal to be compensated. At this time, the signal to be compensated is in analog signal form. The second compensation module 203 can receive the signal to be compensated and the compensation data output by the restoration module 1026 in the auxiliary link. The second compensation module 203 can compensate the signal to be compensated based on the compensation data to output the target signal to the analog signal processing circuit 205. The generation of compensation data in the auxiliary link can be found in subsequent embodiments and will not be described in detail here.

[0066] In some feasible implementations, the acquisition module 1024 can be used to acquire the source data output from the source data output terminal, and then obtain sampled data based on the source data. Here, the source data can be a digital signal or an analog signal, which can be determined according to the actual source data output terminal, and is not limited here. In the transmitter device 20, the source data output terminal can be any signal terminal in the main link other than the second compensation module 203, that is, the source data output terminal can be any signal terminal in the main link other than the second compensation module 203 between the input terminal of the signal generation module 201 and the output terminal of the analog signal processing circuit 205. It can be understood that the source data can be the input signal, output signal, and any signal inside the module of any module other than the second compensation module 203 between the input terminal of the signal generation module 201 and the output terminal of the analog signal processing circuit 205. For example, please refer to Figure 4 ,like Figure 4As shown, the source data output terminal can be the output terminal of the signal generation module 201, and the acquisition module 1024 can acquire the output signal of the signal generation module 201 as the source data, which is a digital signal. In this embodiment, when the device is a transmitter device, any signal terminal between the input terminal of the signal generation module in the main link and the output terminal of the one or more analog signal processing circuits, excluding the second compensation module, can be used as the source data output terminal to output the source data. Here, the signal terminal can be the input terminal of the module, the output terminal of the module, or any signal position inside the module. The source data can be output from any signal terminal between the input terminal of the signal generation module and the output terminal of one or more analog signal processing circuits, excluding the second compensation module, which enriches the sources of source data acquisition in the transmitter device. The acquisition module can flexibly process source data from different locations, and has high applicability.

[0067] In some feasible implementations, the low-frequency calibration module 1025 can acquire a set of filter coefficients, which may include multiple filter coefficients. This set of filter coefficients can be used to filter the sampled data to obtain a differential signal. It is understood that this set of filter coefficients can be directly obtained from a storage file; for details, please refer to the embodiment corresponding to the aforementioned receiver device 10, which will not be elaborated here. Optionally, the set of filter coefficients can be calculated based on a first reference frequency response, a second reference frequency response, and a target frequency response. Here, the first reference frequency response can be the frequency response of the original system from the starting point to the source data output point without compensation by the second compensation module; the second reference frequency response can be the frequency response of the original system from the starting point to the ending point; and the target frequency response can be the desired frequency response of the original system from the starting point to the ending point. The meaning of "original system without compensation by the second compensation module" here is similar to that of "original system without compensation by the first compensation module" mentioned above, and will not be elaborated here.

[0068] Here, in the transmitter device 20, the aforementioned end point is any signal terminal from the input terminal of the aforementioned digital-to-analog converter module 204 to the output terminal of the aforementioned analog signal processing circuit 205, and the aforementioned start point is any signal terminal before the aforementioned end point except for the aforementioned second compensation module 203. It can be understood that the aforementioned end point can be any input terminal, output terminal, or any signal position within any module from the aforementioned digital-to-analog converter module 204 to the output terminal of the aforementioned analog signal processing circuit 205, and the aforementioned start point can be any input terminal, output terminal, or any signal position within any module before the aforementioned end point except for the aforementioned second compensation module 203. The methods for obtaining the first reference frequency response, the second reference frequency response, and the target frequency response, as well as the calculation method of the filter coefficient set, can all be found in the embodiments corresponding to the aforementioned receiver device 10, with differences only in the selectable signal terminals of the start and end points, which will not be elaborated upon here. In the embodiments of this application, the filter coefficient set used in the low-frequency calibration module can be obtained based on the first reference frequency response, the second reference frequency response, and the target frequency response. The first reference frequency response can be the frequency response of the original system from the starting point to the aforementioned source data output point; the second reference frequency response is the frequency response of the original system from the aforementioned starting point to the aforementioned ending point; and the target frequency response is the desired frequency response of the original system from the aforementioned starting point to the aforementioned ending point. The aforementioned ending point can be any one of the aforementioned digital-to-analog conversion module or one or more of the aforementioned analog signal processing circuits, and the aforementioned starting point can be any module preceding the aforementioned ending point except for the aforementioned second compensation module. In other words, both the aforementioned starting point and the aforementioned ending point in the transmitter device can be flexibly selected, specifically determined according to the actual application scenario and the module connection method within the device, thus improving the adaptability of the device. By determining the reference frequency response through the system of various modules in the device, and then calculating the filter coefficient set, the filter coefficient set can be made more adaptable to the actual working state of each module in the device. This allows the filter coefficient set to more accurately reflect the difference between the target amplitude and phase flatness and the amplitude and phase flatness to be compensated. As a result, the difference signal obtained by filtering based on the filter coefficient set is the difference in signal quantity between the signal to be compensated under the target amplitude and phase flatness and the signal quantity to be compensated. Thus, the compensation data generated based on the difference signal can be used to compensate the signal to be compensated, thereby compensating for the signal quantity difference and achieving low-frequency amplitude and phase calibration.

[0069] In some feasible implementations, the second compensation module 203 can combine the compensation data and the signal to be compensated using an adder or combiner. For example, an adder can be used to superimpose the compensation data in digital signal form and the signal to be compensated; alternatively, a combiner can be used to combine the compensation data in analog signal form and the signal to be compensated, thereby compensating for the signal difference between the signal to be compensated under the target amplitude and phase flatness and the amplitude and phase flatness to be compensated, and generating the target signal. The specific implementation can be determined according to the actual application scenario and is not limited here. For example, such as Figure 4 As shown, the signal to be compensated is an analog signal, and the compensation data is also an analog signal. The second compensation module 203 can use a combiner to combine the compensation data and the signal to be compensated in order to compensate for the above signal difference and achieve low-frequency amplitude and phase calibration.

[0070] Example 3

[0071] Based on the above embodiments, in some feasible implementations, in any of the above devices, the first sampling rate of the sampled data output by the acquisition module can be limited to less than or equal to 1 / 2 of the second sampling rate of the source data. It is understood that the first sampling rate can be a value selected based on actual resource status or system status, or a value determined based on historical working conditions; the specific value can be determined according to the actual application scenario and is not limited here. In the embodiments of this application, the first sampling rate can be limited to less than or equal to 1 / 2 of the second sampling rate. When obtaining the differential signal through filtering in the low-frequency calibration module, if an FIR filter is used, the calibration accuracy can generally be considered as the ratio of the sampling rate to the filter order plus one. By implementing data downsampling through the acquisition module, when obtaining the differential signal through filtering in the low-frequency calibration module, the sampling rate of the data after downsampling is lower, which can achieve higher calibration accuracy without changing the filter order. Simultaneously, since the sampling rate of the data after downsampling is lower, a smaller filter order is required to achieve the same calibration accuracy, reducing resource consumption and resource requirements, thus simultaneously addressing resource issues and achieving high accuracy in low-frequency amplitude and phase calibration. Meanwhile, the acquisition module can flexibly set the value of the first sampling rate based on the working status of the device or each module within the device, making the acquisition of sampling data highly flexible, adaptable to changes in the device and resources, and highly applicable.

[0072] In some feasible implementations, in any of the above-mentioned devices, the low-frequency band can be limited to less than or equal to half of the system bandwidth of the device. It is understood that the low-frequency band can be a frequency band starting from 0Hz, and the bandwidth of this band can be less than or equal to half of the system bandwidth of the device. The range of the low-frequency band can be determined based on the actual resource status or system status, or based on historical operating conditions. The specific range can be determined according to the actual application scenario and is not limited here. In the embodiments of this application, the low-frequency band can be limited to less than or equal to half of the system bandwidth. The acquisition module can reduce the processing bandwidth of the source data by downsampling the sampling rate, which can reduce the design order of the filter and the real-time calculation load in the subsequent low-frequency band calibration module, thereby meeting the requirement of reducing resource consumption and resource demand within the device while maintaining the same calibration accuracy. Simultaneously, the acquisition module can flexibly set the value range of the low-frequency band based on the operating status of the device or its various modules, making the acquisition of sampling data highly flexible, adaptable to changes in the device and resources, and highly applicable.

[0073] In some feasible implementations, when the source data is a digital signal, the acquisition module can filter and downsample the source data to reduce its sampling rate from the second sampling rate to the first sampling rate, thereby obtaining sampled data with the first sampling rate. The acquisition module 1024 can use filtering tools such as FIR filters to filter and downsample the source data. For example, an FIR filter can be used for anti-aliasing filtering and downsampling, filtering and decimating the 400 gigabits per second (GSa / s) digital signal output from the amplitude-phase calibration to 195.3125 megasamples per second (MSa / s) of sampled data. The specific filtering and sampling method can be determined according to the actual application scenario and is not limited here. It is understood that the first sampling rate can be limited to less than or equal to half of the second sampling rate; the specific value can be determined according to the actual application scenario and is not limited here. In this embodiment, the source data is a digital signal. The acquisition module can filter and downsample the source data to reduce its sampling rate from a second sampling rate to a first sampling rate, thereby obtaining sampled data with the first sampling rate. Using filtering and downsampling in the acquisition module effectively reduces the introduction of distortion or loss. Reducing the sampling rate of the source data from the second to the first sampling rate in the acquisition module reduces the data processing load in the subsequent low-frequency calibration module. Furthermore, when obtaining the difference signal through filtering in the low-frequency calibration module, the lower sampling rate of the downsampled data results in a smaller filter order required to achieve the same calibration accuracy, thus reducing data processing load, resource consumption, and resource requirements.

[0074] In some feasible implementations, when the source data is an analog signal, the acquisition module can incorporate devices such as a low-pass filter and an analog-to-digital converter (ADC). The source data is low-pass filtered and converted to digital signal using the low-pass filter and ADC to obtain the corresponding digital signal. The acquisition module can then filter and downsample the corresponding digital signal to reduce its sampling rate from a second sampling rate to a first sampling rate, obtaining sampled data with the first sampling rate. Here, the low-pass filter can be a passive filter, an active filter, a switched-capacitor filter, etc. The order and bandwidth of the low-pass filter can be determined according to the actual application scenario, such as based on the first sampling rate requirement and calibration accuracy, and are not limited here. The acquisition module can further filter and downsample the digital signal corresponding to the source data using filtering tools such as FIR filters. For example, an FIR filter can be used to perform anti-aliasing filtering and downsampling on the aforementioned digital signal, filtering and decimating the 400GSa / s digital signal output from the analog-to-digital converter to 195.3125MSa / s sampled data. The specific filtering and sampling method can be determined according to the actual application scenario and is not limited here. It can be understood that the first sampling rate can be limited to less than or equal to half of the second sampling rate; the specific value can be determined according to the actual application scenario and is not limited here. For easier understanding, please refer to [link to relevant documentation]. Figure 3 ,by Figure 3 The receiver device shown is used as an example for explanation. Figure 3As shown, the output of the analog front-end module 1011 serves as the source data output terminal, capable of outputting source data to the acquisition module 1024. This source data is in analog signal form. The acquisition module 1024 may include a low-pass filter 10241, an analog-to-digital converter 10242, and a filtering and decimation submodule 10243. The low-pass filter 10241 performs low-pass filtering on the source data input to the acquisition module 1024 to output low-pass data to the analog-to-digital converter 10242. The analog-to-digital converter 10242 performs analog-to-digital conversion on the low-pass data to output the digital signal corresponding to the source data to the filtering and decimation submodule 10243. The filtering and decimation submodule 10243 filters and decimates the digital signal corresponding to the input source data, reducing the sampling rate of the digital signal to a first sampling rate before outputting sampled data. Optionally, when the source data is an analog signal, the acquisition module can also use a low-pass filter and an analog-to-digital converter (ADC) with an output digital signal sampling rate of the first sampling rate to perform low-pass filtering and analog-to-digital conversion on the source data, generating sampled data with the first sampling rate. Here, the ADC can be selected according to the value of the first sampling rate; that is, the restoration module can introduce an ADC with the first sampling rate parameter, and perform analog-to-digital conversion through this ADC to directly output a digital signal with the first sampling rate as sampled data. In this embodiment, the source data can be an analog signal. The acquisition module can introduce a low-pass filter and an ADC to perform low-pass filtering and analog-to-digital conversion on the source data to obtain the corresponding digital signal, and then filter and downsample the digital signal to generate sampled data with the first sampling rate; or the source data can be directly generated by performing low-pass filtering and analog-to-digital conversion on the source data using a low-pass filter and an ADC with an output digital signal sampling rate of the first sampling rate. Here, performing low-pass filtering on the source data first can avoid aliasing distortion of the signal during analog-to-digital conversion. If low-pass processing is followed by analog-to-digital conversion, and then filtering and downsampling are used to acquire sampled data, distortion or loss can be effectively reduced. This approach also lowers the requirements for the analog-to-digital converter (ADC), allowing for a wider selection of ADCs and greater applicability. Alternatively, if low-pass processing is followed by direct generation of sampled data using a low-sampling-rate ADC, the reconstruction module is simpler, reducing complexity and enriching the methods for acquiring sampled data. Since calibration accuracy can typically be considered as the ratio of the sampling rate to the filter order plus one, reducing the sampling rate of the digital signal corresponding to the source data from the second sampling rate to the first sampling rate in the acquisition module allows for higher calibration accuracy when the difference signal is obtained through filtering in the low-frequency calibration module, as the sampling rate of the downsampled data is lower and the filter order remains unchanged.Meanwhile, since the sampling rate of the downsampled data is lower, the filter order required to achieve the same calibration accuracy is smaller, which reduces resource consumption and resource requirements. It can simultaneously address resource issues and achieve high accuracy in low-frequency amplitude and phase calibration.

[0075] In some feasible implementations, each filter coefficient in the aforementioned filter coefficient group can be a time-domain filter coefficient or a frequency-domain filter coefficient. The low-frequency calibration module can be used to perform time-domain filtering or frequency-domain filtering on the aforementioned sampled data according to the aforementioned filter coefficient group to obtain the aforementioned difference signal. Specifically, when each filter coefficient in the filter coefficient group is a time-domain filter coefficient, since the sampled data input to the low-frequency calibration module is usually a time-domain signal, the low-frequency calibration module can use a time-domain filter (such as an FIR filter) to convolve the difference signal with each filter coefficient to generate the difference signal. When the filter coefficients in the filter coefficient group are frequency domain filter coefficients, since the sampled data input to the low-frequency calibration module is usually a time domain signal, a frequency domain filter can be used in the low-frequency calibration module to perform a fast fourier transform (FFT) on the sampled data to obtain a frequency domain signal. Then, the frequency domain filter coefficients are multiplied, and the result is transformed back to the time domain using an inverse fast fourier transform (IFFT) to obtain the difference signal. It should be understood that the aforementioned time domain filter coefficients and frequency domain filter coefficients can also be converted using FFT or IFFT. That is, time domain filter coefficients can be converted to frequency domain filter coefficients through FFT operations to apply to a frequency domain filter, and frequency domain filter coefficients can be converted to time domain filter coefficients through IFFT operations to apply to a time domain filter; this is not a limitation. In this embodiment, the low-frequency calibration module can perform time-domain filtering or frequency-domain filtering on the sampled data based on the filter coefficient set to obtain the difference signal. The filter coefficient set and filtering tools can be flexibly selected to implement the above filtering process, which improves the flexibility of the device and makes it highly applicable.

[0076] In some feasible implementations, when both the source data and the signal to be compensated are digital signals, the restoration module can perform direct interpolation on the difference signal or perform filtered interpolation on the difference signal to convert the sampling rate of the difference signal to the third sampling rate of the signal to be compensated, thereby generating first interpolated data. The low-frequency portion of the first interpolated data is consistent with the low-frequency portion of the difference signal, and the energy of other frequency components in the first interpolated data, excluding the low-frequency portion, is suppressed. The restoration module can then output the first interpolated data as compensation data. It is understood that the third sampling rate of the signal to be compensated may or may not be consistent with the second sampling rate of the source data, determined by the signal processing from the source data output to the input of the first or second compensation module in the main link, and is not limited here. It is understood that the restoration module can perform direct interpolation or filtered interpolation on the difference signal to convert the sampling rate of the difference signal to the third sampling rate of the signal to be compensated. This ensures that the sampling rate of the generated compensation data matches the sampling rate of the signal to be compensated, allowing the compensation data to be merged into the signal to be compensated. Here, keeping the low-frequency portion of the first interpolated data consistent with the low-frequency portion of the difference signal can be understood as the low-frequency portions being approximately the same or approximately within an acceptable range. This can be determined based on the actual application scenario and is not limited here. The restoration module can insert approximately zero values ​​into the frequency portions of the difference signal (excluding the low-frequency portion) through direct interpolation or filtered interpolation, so that the sampling rate of the generated first interpolated data is converted to the third sampling rate and the energy of other frequency portions in the first interpolated data is suppressed. In this embodiment, when both the source data and the compensation signal are digital signals, the restoration module can perform direct interpolation or filtered interpolation on the difference signal to convert the sampling rate of the difference signal to the third sampling rate of the signal to be compensated to generate first interpolated data, which can then be output as compensation data. Here, the low-frequency component in the first interpolated data is consistent with the low-frequency component in the differential signal, and the energy of other frequency components in the first interpolated data, excluding the low-frequency component, is suppressed. This avoids introducing interference to the subsequent low-frequency amplitude and phase compensation of the signal to be compensated, thus improving the accuracy of amplitude and phase calibration. Through interpolation processing, the sampling rate of the compensation data can be converted to the third sampling rate of the signal to be compensated, allowing the compensation data to be directly merged into the signal to be compensated with the same sampling rate. This compensates for the signal quantity difference between the signal to be compensated under the target amplitude and phase flatness and under the amplitude and phase flatness to be compensated, thereby achieving low-frequency amplitude and phase calibration.

[0077] In some feasible implementations, when the source data is a digital signal and the signal to be compensated is an analog signal, the restoration module can be used to perform digital-to-analog conversion on the difference signal to obtain a first converted signal, perform synchronous calibration on the first converted signal, and output the synchronously calibrated signal as compensation data; or perform direct interpolation processing on the difference signal, or perform filtered interpolation processing on the difference signal to increase the sampling rate of the difference signal to generate second interpolated data, wherein the low-frequency part of the second interpolated data is consistent with the low-frequency part of the difference signal, and the energy of other frequency parts in the second interpolated data excluding the low-frequency part is suppressed; perform digital-to-analog conversion on the second interpolated data to obtain a second converted signal, perform synchronous calibration on the second converted signal, and output the synchronously calibrated signal as compensation data. It is understandable that there may be link delays between the main link and the auxiliary link in the device. For example, when the source data is a digital signal and the signal to be compensated is an analog signal, there may be link delays between the two links due to signal processing delays, differences in analog hardware, differences in transmission links, etc. For example, the analog hardware differences introduced into the auxiliary link due to the digital-to-analog converter can cause the compensation data output from the auxiliary link and the signal to be compensated transmitted from the main link to arrive at the first / second compensation module asynchronously in time. Therefore, it is necessary to introduce synchronization calibration in the restoration module (such as increasing the delay of the signal passing through the auxiliary link) to compensate for the link delay between the two links. Here, synchronization calibration can be achieved through analog domain devices such as numerically controlled analog delay lines and voltage-controlled delay units. For example, due to the large link delay caused by the signal processing delay of the main link, the delay of the auxiliary link in transmitting compensation data can be increased by numerically controlled analog delay lines so that the signal to be compensated and the compensation data are synchronized when they arrive at the first / second compensation module, that is, to compensate for the link delay between the two links to achieve time synchronization between the signal to be compensated and the compensation signal. It is understood that the aforementioned link delay can be stored / configured within the analog domain device of the restoration module. The restoration module can use the stored / configured link delay parameters within the device to synchronously calibrate the first or second converted signal to compensate for the link delay, ensuring that the generated compensation data is synchronized with the signal to be compensated in time. Optionally, when the internal modules of the main link and auxiliary link in the device are connected by PCB traces, synchronous calibration can also be achieved by adjusting the PCB trace length of the auxiliary link to compensate for the link delay between the two links, ensuring that the generated compensation data is synchronized with the signal to be compensated in time. Here, the specific implementation method of synchronous calibration can be determined according to the actual application scenario and is not limited here.It is understandable that the restoration module can directly output the signal obtained by digital-to-analog conversion and synchronous calibration of the difference signal as the compensation data, or it can first perform interpolation processing on the difference signal (such as the direct interpolation processing or the filtered interpolation processing mentioned above) to increase the sampling rate of the difference signal, generating second interpolated data. Then, the signal obtained by digital-to-analog conversion and synchronous calibration of the second interpolated data is used as the compensation data output. Here, the sampling rate of the second interpolated data can be the second sampling rate of the source data; the specific sampling rate value can be determined according to the actual application scenario and is not limited here. It is understandable that upsampling through interpolation can improve the signal-to-noise ratio of the generated compensation data to a certain extent, reduce noise, and minimize the introduction of undesirable factors. For easier understanding, please refer to [link to relevant documentation]. Figure 5 ,like Figure 5As shown, the signal to be compensated is the signal output by the digital-to-analog converter module 204 in the main link, and this signal is in analog signal form. The restoration module 1026 may include an interpolation processing submodule 10261, a digital-to-analog converter 10262, and a synchronous calibration submodule 10263. It can be understood that the interpolation processing submodule 10261 is an optional module. The interpolation processing submodule 10261 can directly interpolate or filter interpolate the input difference signal to generate second interpolated data with a third sampling rate, and output it to the digital-to-analog converter 10262. The digital-to-analog converter 10262 can perform digital-to-analog conversion on the second interpolated data to generate converted data in analog signal form, and output it to the synchronous calibration submodule 10263. The synchronous calibration submodule 10263 can synchronously calibrate the converted data to output compensated data to the second compensation module 203. It is understood that when the interpolation processing submodule 10261 is absent, the differential signal output by the low-frequency calibration module 1025 can be directly input into the digital-to-analog converter 10262 to be converted into analog signal form and output as synchronous calibration submodule 10263. In this embodiment, when the source data is a digital signal and the signal to be compensated is an analog signal, the restoration module can directly perform digital-to-analog conversion on the differential signal to generate a first converted signal, and output the signal after synchronous processing of the first converted signal as compensation data. Alternatively, the sampling rate of the differential signal can be increased to the third sampling rate of the signal to be compensated to obtain second interpolation data, and then the second interpolation data can be performed digital-to-analog conversion to generate a second converted signal, and output the signal after synchronous calibration of the second converted signal as compensation data. It is understood that the compensation data generated above is in analog signal form. Synchronous calibration can compensate for the link delay between the different signal links where the signal to be compensated and the compensation data are located, so that the compensation data and the signal to be compensated are synchronized in time. The compensation data can be directly merged with the signal to be compensated in the first / second compensation module, simplifying the merging process of the signal to be compensated and the compensation data in the subsequent first / second compensation modules. This avoids the need to introduce digital-to-analog conversion or synchronization processing in the first / second compensation modules. The first / second compensation modules can directly combine the two analog signals through combiners to generate the target signal, reducing the complexity of the first / second compensation modules. At the same time, it enriches the way compensation data is generated, and can flexibly generate compensation data according to the application scenario and resource status, making it highly applicable.

[0078] In some feasible implementations, when the source data is an analog signal and the signal to be compensated is a digital signal, the restoration module can be used to perform direct interpolation processing on the difference signal or to perform filtered interpolation processing on the difference signal to convert the sampling rate of the difference signal to a third sampling rate of the signal to be compensated to generate third interpolated data. The low-frequency component of the third interpolated data is consistent with the low-frequency component of the difference signal, and the energy of other frequency components in the third interpolated data, excluding the low-frequency component, is suppressed. Furthermore, the restoration module can be used to perform synchronous calibration on the third interpolated data, and output the data after synchronous calibration as the compensation data. It is understandable that there may be link delays between the main link and the auxiliary link in the device. For example, when the source data is an analog signal and the signal to be compensated is a digital signal, there may be link delays between the two links caused by signal processing delays, differences in analog hardware, differences in transmission links, and digital acquisition deviations. If there are many signal processing modules in the main link, the signal processing delay may be large, increasing the signal transmission delay through the main link. Synchronization processing can be used to increase the signal transmission delay through the auxiliary link to compensate for the link delay between the two links. It is understood that the aforementioned link delays can be stored in memory. The restoration module can read the link delays from memory for calculations or directly read the calculated parameters and perform synchronous calibration on the third interpolation data to compensate for the link delays, ensuring that the generated compensation data is synchronized with the signal to be compensated in time. Here, synchronous calibration can be implemented using digital domain modules such as adaptive filters and fractional delay filters. For example, the link delay caused by the large signal processing delay of the main link can be compensated by inputting the compensation data into the aforementioned digital domain module to increase the delay of the auxiliary link, thereby compensating for the link delay. It is understood that the aforementioned digital domain module can be implemented using an FPGA or a CPU, depending on the specific application scenario, and no restrictions are imposed here. For example, please refer to [link to example]. Figure 6 , Figure 6 This is another data interaction diagram in the receiver device provided in the embodiments of this application. For example... Figure 6 As shown, the auxiliary link acquisition module 1024 incorporates a low-pass filter 10241 and an analog-to-digital converter 10242, which may lead to the aforementioned differences in analog hardware, resulting in link delay between the main link and the auxiliary link. Therefore, synchronization calibration is required to compensate for the aforementioned link delay. Figure 6As shown, the restoration module 1026 may include an interpolation processing submodule 10261 and a synchronization calibration submodule 10263. The interpolation processing submodule 10261 can directly interpolate or filter interpolate the input difference signal to generate third interpolated data with a third sampling rate. The synchronization calibration submodule 10263 can retrieve the aforementioned link delay from memory and perform synchronous calibration on the third interpolated data based on the link delay to compensate for the link delay and generate compensated data. It can be understood that the compensated data output by the synchronization calibration submodule 10263 is approximately synchronized with the signal to be compensated in time, and the compensated data can be directly superimposed on the signal to be compensated to achieve amplitude and phase calibration. Optionally, the order of the interpolation processing and the synchronous calibration described above can be reversed. That is, in the restoration module, the difference signal can be synchronously calibrated first to obtain a calibration signal, and then the calibration signal can be directly interpolated or filtered interpolated to convert the sampling rate of the calibration signal to the third sampling rate of the signal to be compensated, thereby generating and outputting compensation data. The specific implementation process of interpolation processing and synchronous calibration can be found in the description of this embodiment, and will not be elaborated here. In this embodiment, when the source data is an analog signal and the signal to be compensated is a digital signal, the restoration module can perform direct interpolation or filtered interpolation on the difference signal to convert the sampling rate of the difference signal to the third sampling rate of the signal to be compensated, thereby generating third interpolated data. Then, the third interpolated data can be synchronously calibrated to output compensation data. Alternatively, the restoration module can perform synchronous calibration on the difference signal and then perform direct interpolation or filtered interpolation to obtain compensation data. Direct interpolation or filtered interpolation can improve the sampling rate of the data, making the sampling rate of the generated compensation data consistent with the sampling rate of the signal to be compensated, which can be directly merged with the signal to be compensated to achieve amplitude and phase calibration. The low-frequency component of the compensation data is consistent with the low-frequency component of the interpolated data, and the energy of other frequency components in the third interpolated data, excluding the low-frequency component, is suppressed, avoiding the introduction of interference in subsequent compensation. Simultaneously, synchronous calibration processing can compensate for the link delay between the signal to be compensated and the compensation data located in different signal links, allowing the generated compensation data to be directly superimposed on the signal to be compensated, avoiding the introduction of secondary distortion. This further improves the adaptability of the device, enabling it to adapt to complex signal processing processes and demonstrating high applicability.

[0079] In summary, this application discloses an amplitude and phase calibration apparatus, which can be a receiver or a transmitter. The receiver may include an analog front-end module, an analog-to-digital conversion module, multiple first calibration modules, a first compensation module, and one or more digital signal processing modules. The transmitter may include a signal generation module, multiple second calibration modules, a second compensation module, a digital-to-analog conversion module, and one or more analog signal processing circuits. Both the receiver and transmitter include an acquisition module, a low-frequency calibration module, and a restoration module. The acquisition module can acquire source data output from a source data output terminal and acquire sampled data with a first sampling rate based on the source data, where the first sampling rate is lower than the second sampling rate of the source data. The low-frequency calibration module can filter the sampled data according to a set of filtering coefficients to obtain a difference signal, where the difference signal can be the difference in signal magnitude between the signal to be compensated under the target amplitude and phase flatness and the signal under the amplitude and phase flatness to be compensated. In the receiver, the signal to be compensated can be any signal between the input terminal of the analog front-end module and the output terminal of the one or more digital signal processing modules. In the transmitter device, the signal to be compensated can be any signal between the input terminal of the aforementioned signal generation module and the output terminal of the aforementioned one or more analog signal processing circuits, allowing for high flexibility in the selection of the signal to be compensated. Since calibration accuracy can generally be considered as the ratio of the sampling rate to the filter order plus one, the acquisition module implements data downsampling. When obtaining the differential signal through filtering in the low-frequency calibration module, the sampling rate of the downsampled data is lower, achieving higher calibration accuracy without changing the filter order. Simultaneously, due to the lower sampling rate of the downsampled data, a smaller filter order is required to achieve the same calibration accuracy, reducing resource consumption and requirements. This approach simultaneously addresses resource constraints and achieves high accuracy in low-frequency amplitude and phase calibration. The aforementioned restoration module can output compensation data that matches the type of the signal to be compensated based on the differential signal. Here, the compensation data can be merged into the signal to be compensated to achieve amplitude and phase calibration. Using the device provided in this application, high-precision low-frequency amplitude and phase calibration can be achieved to improve the low-frequency amplitude and phase flatness of the system. The device has a simple structure and high flexibility in the connection between its modules, making it highly applicable.

[0080] It is understood that each module mentioned in the above embodiments can be implemented by software, hardware, or a combination of software and hardware, for example, as Figure 1a In the illustrated embodiment, the aforementioned plurality of first calibration modules, first compensation module 1022, digital signal processing module 1023, acquisition module 1024, low-frequency band calibration module 1025, and restoration module 1026 in receiver device 10 can all be implemented based on software.

[0081] In this application, "implemented in software" means that the processor reads and executes data stored in memory (such as...). Figure 1a The processor implements the functions corresponding to the above modules or units using program instructions stored in memory 103. Here, a processor refers to a processing circuit capable of executing program instructions, including but not limited to at least one of the following: CPU, microprocessor, digital signal processor (DSP), microcontroller unit (MCU), or artificial intelligence processor, etc., and other processing circuits capable of running program instructions. In other embodiments, the processor may also include circuits with other processing functions (such as hardware circuits for hardware acceleration, bus and interface circuits, etc.). The processor can be presented as an integrated chip, for example, as an integrated chip whose processing function only includes executing software instructions, or it can also be presented as a SoC (system on a chip), that is, on a single chip, in addition to the processing circuit capable of running program instructions (usually referred to as the "core"), it also includes other hardware circuits for implementing specific functions (of course, these hardware circuits can also be implemented separately based on ASIC or FPGA). Correspondingly, the processing function includes not only executing software instructions but also various hardware acceleration functions (such as AI calculation, encoding / decoding, compression / decompression, etc.).

[0082] In this application, "implemented in hardware" means that the functions of the above-mentioned modules or units are implemented through hardware processing circuits that do not have program instruction processing capabilities. These hardware processing circuits can be composed of discrete hardware components or integrated circuits. To reduce power consumption and size, integrated circuits are typically used. The hardware processing circuits can include application-specific integrated circuits (ASICs) or programmable logic devices (PLDs); PLDs can include FPGAs, complex programmable logic devices (CPLDs), and so on. These hardware processing circuits can be a single packaged semiconductor chip (e.g., packaged as an ASIC); or they can be integrated with other circuits (e.g., CPUs, DSPs) and packaged into a single semiconductor chip. For example, multiple hardware circuits and a CPU can be formed on a silicon substrate and packaged into a single chip, also known as a System-on-a-Chip (SoC). Alternatively, circuits for implementing FPGA functions and a CPU can be formed on a silicon substrate and packaged into a single chip, also known as a System-on-a-Chip (SoPC).

[0083] It should be noted that when this application is implemented through software, hardware, or a combination of both, different software or hardware can be used, and it is not limited to using only one type of software or hardware. For example, one module or unit can be implemented using a CPU, while another module or unit can be implemented using a DSP. Similarly, when implemented using hardware, one module or unit can be implemented using an ASIC, while another module or unit can be implemented using an FPGA. Of course, it is not limited to using the same software (e.g., all through a CPU) or the same hardware (e.g., all through an ASIC) to implement some or all modules or units. Furthermore, those skilled in the art will understand that software is generally more flexible but less performant than hardware, while hardware is the opposite. Therefore, those skilled in the art can choose software, hardware, or a combination of both based on actual needs.

[0084] The term "comprising," and any variations thereof, in the specification, claims, and drawings of this application are intended to cover a non-exclusive inclusion. For example, a process, method, apparatus, product, or device that includes a series of steps or modules is not limited to the steps or modules listed, but may optionally include steps or modules not listed, or may optionally include other steps or modules inherent to such process, method, apparatus, product, or device.

[0085] Those skilled in the art will recognize that the modules and algorithm steps of the various examples described in conjunction with the embodiments disclosed herein can be implemented in electronic hardware, computer software, or a combination of both. To clearly illustrate the interchangeability of hardware and software, the components and steps of the various examples have been generally described in terms of functionality in the foregoing description. Whether these functions are implemented in hardware or software depends on the specific application and design constraints of the technical solution. Those skilled in the art can use different methods to implement the described functions for each specific application, but such implementation should not be considered beyond the scope of this application.

[0086] The above-described preferred embodiments have further detailed the purpose, technical solutions, and advantages of this application. It should be understood that the above are merely embodiments of this application and are not intended to limit this application. Any modifications, equivalent substitutions, improvements, etc., made within the spirit and principles of this application should be included within the protection scope of this application.

Claims

1. An amplitude and phase calibration device, wherein the device is a receiver device or a transmitter device, characterized in that: The receiver device includes an analog front-end module, an analog-to-digital conversion module, multiple first calibration modules, a first compensation module, and one or more digital signal processing modules. The analog front-end module is used to output a first start signal to the analog-to-digital conversion module. The analog-to-digital conversion module is used to perform analog-to-digital conversion on the first start signal to obtain a digital calibration signal and output it to the multiple first calibration modules. The multiple first calibration modules are used to perform calibration processing on the digital calibration signal and output a calibrated first processed signal to the one or more digital signal processing modules. The one or more digital signal processing modules are used to perform signal processing on the first processed signal. The transmitter device includes a signal generation module, multiple second calibration modules, a second compensation module, a digital-to-analog converter module, and one or more analog signal processing circuits. The signal generation module is used to output a second start signal to the multiple second calibration modules. The multiple second calibration modules are used to perform calibration processing on the second start signal and output a second processed signal to the digital-to-analog converter module. The digital-to-analog converter module is used to perform digital-to-analog conversion on the second processed signal to obtain an analog calibration signal and output it to the one or more analog signal processing circuits. The one or more analog signal processing circuits are used to perform signal processing on the analog calibration signal. The device also includes an acquisition module, a low-frequency calibration module, and a restoration module; The acquisition module is used to acquire the source data output by the source data output terminal, and obtain sampled data based on the source data, wherein the first sampling rate of the sampled data is less than the second sampling rate of the source data; The low-frequency calibration module is used to filter the sampled data according to the filter coefficient set to obtain a difference signal. The filter coefficient set includes multiple filter coefficients. The filter coefficient set is used to reflect the difference between the target amplitude-phase flatness and the amplitude-phase flatness to be compensated of the system. The difference signal is the difference in signal quantity between the signal to be compensated under the target amplitude-phase flatness and the amplitude-phase flatness to be compensated. Wherein, when the device is the receiver device, the signal to be compensated is any signal between the input terminal of the analog front-end module and the output terminal of the one or more digital signal processing modules; when the device is the transmitter device, the signal to be compensated is any signal between the input terminal of the signal generation module and the output terminal of the one or more analog signal processing circuits. The restoration module is used to output compensation data that matches the type of the signal to be compensated based on the difference signal. The compensation data is used to merge into the signal to be compensated to achieve amplitude and phase calibration. Both the first compensation module and the second compensation module are used to compensate the signal to be compensated based on the compensation data to generate a target signal after amplitude and phase calibration.

2. The apparatus according to claim 1, characterized in that, The filter coefficient set consists of multiple coefficients obtained based on a first reference frequency response, a second reference frequency response, and a target frequency response. The first reference frequency response is the frequency response of the original system from the starting point to the source data output point. The second reference frequency response is the frequency response of the original system from the starting point to the ending point. The target frequency response is the desired frequency response of the original system from the starting point to the ending point. The original system is a system that has not been compensated by the compensation module. When the device is a receiver device, the compensation module is the first compensation module, the starting end is any signal end from the input end of the analog front-end module to the output end of the analog-to-digital conversion module, and the ending end is any signal end after the starting end except for the first compensation module. When the device is a transmitter device, the compensation module is the second compensation module, the end terminal is any signal terminal from the input terminal of the digital-to-analog conversion module to the output terminal of the one or more analog signal processing circuits, and the start terminal is any signal terminal other than the second compensation module before the end terminal.

3. The apparatus according to claim 2, characterized in that, The filter coefficient set consists of multiple coefficients obtained based on the difference between the target frequency response and the second reference frequency response, as well as the first reference frequency response.

4. The apparatus according to any one of claims 1 to 3, characterized in that, Each filter coefficient in the filter coefficient group is a time-domain filter coefficient or a frequency-domain filter coefficient. The low-frequency calibration module, when filtering the sampled data according to the filter coefficient group to obtain the difference signal, is specifically used for: The sampled data is subjected to time-domain filtering or frequency-domain filtering based on the filtering coefficient set to obtain the difference signal.

5. The apparatus according to any one of claims 1 to 3, characterized in that, The first sampling rate is less than or equal to half of the second sampling rate.

6. The apparatus according to any one of claims 1 to 3, characterized in that, The low-frequency band is less than or equal to 1 / 2 of the system bandwidth of the device.

7. The apparatus according to claim 2 or 3, characterized in that, The device is a receiver device, and the source data output terminal is any signal terminal in the main link except for the first compensation module. The main link consists of the analog front-end module, the analog-to-digital conversion module, the plurality of first calibration modules, the first compensation module, and the one or more digital signal processing modules.

8. The apparatus according to claim 2 or 3, characterized in that, The device is a transmitter device, and the source data output terminal is any signal terminal in the main link except for the second compensation module. The main link consists of the signal generation module, the plurality of second calibration modules, the second compensation module, the digital-to-analog conversion module, and the one or more analog signal processing circuits.

9. The apparatus according to any one of claims 1 to 3, characterized in that, The source data is a digital signal, and the acquisition module is used for: The source data is filtered and downsampled to reduce the sampling rate of the source data from the second sampling rate to the first sampling rate, thereby obtaining sampled data with the first sampling rate.

10. The apparatus according to claim 9, characterized in that, When the signal to be compensated is a digital signal, the restoration module is used to: The difference signal is directly interpolated or filtered and interpolated to convert the sampling rate of the difference signal to the third sampling rate of the signal to be compensated to generate first interpolated data. The low-frequency part of the first interpolated data is consistent with the low-frequency part of the difference signal, and the energy of other frequency parts in the first interpolated data other than the low-frequency part is suppressed. The first interpolated data is output as compensation data.

11. The apparatus according to claim 9, characterized in that, When the signal to be compensated is an analog signal, the restoration module is used to: The difference signal is converted from digital to analog to obtain a first converted signal, the first converted signal is synchronously calibrated, and the synchronously calibrated signal is output as compensation data; or The difference signal is subjected to direct interpolation or filtered interpolation to increase the sampling rate of the difference signal to generate second interpolated data. The low-frequency component of the second interpolated data is consistent with the low-frequency component of the difference signal, and the energy of other frequency components in the second interpolated data excluding the low-frequency component is suppressed. The second interpolated data is then converted from digital to analog to obtain a second converted signal. The second converted signal is then synchronously calibrated, and the synchronously calibrated signal is output as compensation data.

12. The apparatus according to any one of claims 1 to 3, characterized in that, The source data is an analog signal, and the acquisition module is used for: The source data is subjected to low-pass filtering and analog-to-digital conversion using a low-pass filter and an analog-to-digital converter to obtain the digital signal corresponding to the source data; the digital signal corresponding to the source data is then filtered and downsampled to reduce the sampling rate of the digital signal corresponding to the source data from a second sampling rate to a first sampling rate, thereby obtaining sampled data with the first sampling rate. or The source data is subjected to low-pass filtering and analog-to-digital conversion using a low-pass filter and an analog-to-digital converter with an output digital signal sampling rate of the first sampling rate to obtain the digital signal corresponding to the source data as sampled data, wherein the sampling rate of the sampled data is the first sampling rate.

13. The apparatus according to claim 12, characterized in that, When the signal to be compensated is a digital signal, the restoration module is used to: The difference signal is subjected to direct interpolation or filtered interpolation to convert the sampling rate of the difference signal to a third sampling rate of the signal to be compensated, thereby generating third interpolated data. The low-frequency component of the third interpolated data is consistent with the low-frequency component of the difference signal, and the energy of other frequency components in the third interpolated data, excluding the low-frequency component, is suppressed. The third interpolated data is then synchronously calibrated, and the calibrated data is output as compensation data. The difference signal is synchronously calibrated to obtain a calibration signal; the calibration signal is directly interpolated or filtered and interpolated to convert the sampling rate of the calibration signal to the third sampling rate of the signal to be compensated to generate and output compensation data, wherein the low-frequency part of the compensation data is consistent with the low-frequency part of the calibration signal, and the energy of other frequency parts in the compensation data other than the low-frequency part is suppressed.

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