Instrument frequency conversion method and system based on direct digital frequency synthesis

Abstract

The present application relates to the technical field of instrument frequency conversion, and discloses an instrument frequency conversion method and system based on direct digital frequency synthesis, the method comprising: receiving a target frequency instruction and a reference clock signal, converting the target frequency instruction and the reference clock signal into a phase accumulation step length via a frequency tuning word; mapping a frequency spectrum relationship between the target frequency and a reference clock frequency to obtain a filter control parameter; generating a phase sequence according to the phase accumulation step length and the reference clock signal, and performing coordinate rotation vector iteration on a current phase value in the phase sequence to obtain a digital sinusoidal sequence; performing pulse amplitude modulation on the digital sinusoidal sequence to obtain a stepped analog signal; reconstructing a filter characteristic of the filter control parameter to obtain a current filter passband, and filtering out high-order harmonic and image frequency components in the stepped analog signal to obtain a pure analog signal; and the present application can improve the efficiency of instrument frequency conversion based on direct digital frequency synthesis.

Description

Technical Field

[0001] This invention relates to the field of instrument frequency conversion technology, and in particular to an instrument frequency conversion method and system based on direct digital frequency synthesis. Background Technology

[0002] In existing instrument frequency conversion technologies based on direct digital frequency synthesis, the calculation of the phase accumulation step size often lacks verification of the stability of the reference clock frequency and precise alignment of the bit width, resulting in insufficient accuracy in phase sequence generation, which in turn affects the integrity and accuracy of subsequent digital sine sequences. Simultaneously, the determination of filter control parameters often relies on fixed filter types and parameter ranges, failing to fully integrate the dynamic adaptation of the spectral relationship between the target frequency and the reference clock frequency. This results in weak harmonic suppression and difficulty in meeting the signal processing needs of complex spectral scenarios.

[0003] Existing technologies lack optimal fitting of the passband boundary and transition band slope in the filter characteristic reconstruction stage, resulting in low matching between the filter passband and the target frequency. This makes it difficult to effectively filter out high-order harmonics and image frequency components in stepped analog signals. Furthermore, the identification of spectral positions and the configuration of filter parameters during image frequency suppression are not precise enough, leading to high signal distortion rates and insufficient continuity and efficiency of the overall frequency conversion process. This restricts the overall performance of the instrument's frequency conversion. Therefore, improving the frequency conversion efficiency of instruments based on direct digital frequency synthesis has become an urgent problem to be solved. Summary of the Invention

[0004] This invention provides an instrument frequency conversion method and system based on direct digital frequency synthesis to solve the problems mentioned in the background art.

[0005] To achieve the above objectives, the present invention provides an instrument frequency conversion method based on direct digital frequency synthesis, comprising:

[0006] S1. Receive the target frequency command and reference clock signal from the instrument and equipment, and perform frequency tuning word conversion on the target frequency command according to the reference clock frequency in the reference clock signal to obtain the phase accumulation step size of the instrument and equipment.

[0007] S2. Perform spectral feature mapping on the spectral relationship between the target frequency and the reference clock frequency in the target frequency command to obtain the filtering control parameters of the instrument.

[0008] S3. Based on the phase accumulation step size and the reference clock signal, generate the phase sequence of the instrument and equipment, and perform coordinate rotation vector iteration on the current phase value in the phase sequence to obtain the digital sine sequence of the instrument and equipment;

[0009] S4. Pulse amplitude modulation is performed on the digital sine sequence to obtain the stepped analog signal of the instrument.

[0010] S5. Reconstruct the filtering characteristics of the filtering control parameters to obtain the current filtering passband that matches the target frequency;

[0011] S6. Based on the current filter passband, filter out the higher harmonics and image frequency components in the stepped analog signal to obtain the pure analog signal of the instrument.

[0012] In a preferred embodiment, the step of receiving the target frequency command and reference clock signal from the receiving instrument, and performing frequency tuning word conversion on the target frequency command based on the reference clock frequency in the reference clock signal to obtain the phase accumulation step size of the instrument, includes:

[0013] Receive the target frequency command and reference clock signal from the instrument and equipment, and parse the command format of the target frequency command to obtain the standardized frequency control word of the instrument and equipment;

[0014] The frequency stability of the reference clock frequency in the reference clock signal is verified to obtain the verified reference clock frequency.

[0015] The frequency ratio of the standardized frequency control word to the verified reference clock frequency is calculated to obtain the frequency tuning ratio of the instrument.

[0016] The frequency tuning ratio is bit-width aligned to obtain the phase accumulation step size of the instrument.

[0017] In a preferred embodiment, the step of mapping the spectral relationship between the target frequency in the target frequency command and the reference clock frequency to obtain the filter control parameters of the instrument includes:

[0018] Spectral analysis is performed on the target frequency and the reference clock frequency in the target frequency command to obtain the spectral relationship characteristics of the instrument.

[0019] Based on the spectral relationship characteristics, identify the harmonic suppression requirements of the target frequency at the reference clock frequency;

[0020] The harmonic suppression requirement characteristics are mapped to a preset filter type library to obtain the target filter type of the instrument.

[0021] By matching the target filter type with the harmonic suppression requirement characteristics through parameter derivation, the filter parameter range of the instrument is obtained.

[0022] The filter parameter range is optimized and constrained to obtain the filter control parameters of the instrument.

[0023] In a preferred embodiment, the formula for calculating the cutoff frequency in the filter control parameters is as follows:

[0024] ;

[0025] In the formula, The cutoff frequency, The target frequency, This is the preset passband reference scaling factor. The preset harmonic suppression dynamic adjustment factor, The reference clock frequency is, The highest harmonic order in the harmonic suppression requirement characteristics. It is the natural logarithm function.

[0026] In a preferred embodiment, the step of generating a phase sequence of the instrument based on the phase accumulation step size and the reference clock signal, and performing coordinate rotation vector iteration on the current phase value in the phase sequence to obtain a digital sine sequence of the instrument, includes:

[0027] The phase value of the phase accumulator in the instrument is initialized to zero, and triggered by the clock edge of the reference clock signal, the phase accumulation step size is iteratively superimposed with the real-time phase value of the phase accumulator to obtain the phase sequence of the instrument.

[0028] Based on the bit width of the phase accumulation step size, the current phase value in the phase sequence is truncated to obtain the phase angle to be iterated for the current phase value;

[0029] Based on a preset initial rotation vector, the phase angle to be iterated is successively rotated to approximate it, and the convergent rotation coordinates of the phase angle to be iterated are obtained.

[0030] The convergent rotating coordinates are orthogonally decomposed and extracted to obtain the digital sine sequence of the instrument.

[0031] In a preferred embodiment, the step of pulse amplitude modulation of the digital sine sequence to obtain the stepped analog signal of the instrument includes:

[0032] The digital sample values ​​in the digital sine sequence are converted from digital to analog to obtain the analog level values ​​of the digital sample values;

[0033] Based on the timing of the reference clock signal, the analog level value is time-driven to obtain a constant amplitude pulse of the analog level value;

[0034] The constant amplitude pulse is reconstructed to obtain the pulse amplitude modulation signal of the constant amplitude pulse;

[0035] The pulse amplitude modulation signal is time-domain waveform preserved to obtain the stepped analog signal of the instrument.

[0036] In a preferred embodiment, the step of reconstructing the filter characteristics of the filter control parameters to obtain a current filter passband that matches the target frequency includes:

[0037] Extract the cutoff frequency and filter order from the filter control parameters to generate the filter structure parameters of the instrument.

[0038] Based on the filter structure parameters, the amplitude response and phase response of the filter in the instrument are parametrically reconstructed to obtain the reconstructed frequency response characteristics of the instrument.

[0039] Based on the target frequency and the reconstructed frequency response characteristics, the passband boundary and transition band slope of the filter are optimally fitted to obtain the current filter passband that matches the target frequency.

[0040] In a preferred embodiment, the step of filtering out higher harmonics and image frequency components from the stepped analog signal based on the current filter passband to obtain a clean analog signal for the instrument includes:

[0041] Based on the current passband range and stopband attenuation requirements, configure the center frequency and bandwidth parameters for the filter in the instrument.

[0042] The stepped analog signal is input to a pre-configured filter for bandpass filtering to obtain the preliminary analog signal of the instrument.

[0043] Based on the relative relationship between the target frequency and the reference clock frequency, the preliminary analog signal is subjected to image frequency suppression to obtain the suppressed analog signal of the instrument.

[0044] The total harmonic distortion of the suppressed analog signal is optimized to obtain a clean analog signal from the instrument.

[0045] In a preferred embodiment, the step of performing image frequency suppression on the preliminary analog signal based on the relative relationship between the target frequency and the reference clock frequency to obtain the suppressed analog signal of the instrument includes:

[0046] Based on the target frequency and the reference clock frequency, the spectral characteristics of the stepped analog signal are mapped and identified to obtain the underlying coordinates of the potential mirror frequency of the stepped analog signal.

[0047] Based on the frequency multiplication relationship of the reference clock frequency, the potential mirror frequency base coordinates are multiplied and extended to obtain the specific spectral location set of the potential mirror frequency base coordinates;

[0048] Based on the specific spectral location set, the configuration parameters of the image frequency suppression filter in the instrument are generated in a regularized manner to obtain the stopband center frequency and stopband width parameters of the image frequency suppression filter.

[0049] By applying the stopband center frequency and the stopband width parameters, the preliminary analog signal is band-stop filtered to obtain the suppressed analog signal of the instrument.

[0050] To address the aforementioned problems, the present invention also provides an instrument frequency conversion system based on direct digital frequency synthesis, the system comprising:

[0051] The frequency tuning word conversion module is used to receive the target frequency command and reference clock signal of the instrument and equipment, and to convert the target frequency command into a frequency tuning word according to the reference clock frequency in the reference clock signal, so as to obtain the phase accumulation step size of the instrument and equipment.

[0052] The spectrum feature mapping module is used to perform spectrum feature mapping on the spectrum relationship between the target frequency and the reference clock frequency in the target frequency command to obtain the filtering control parameters of the instrument.

[0053] A digital sine sequence generation module is used to generate a phase sequence of the instrument based on the phase accumulation step size and the reference clock signal, and to perform coordinate rotation vector iteration on the current phase value in the phase sequence to obtain a digital sine sequence of the instrument.

[0054] A pulse amplitude modulation module is used to perform pulse amplitude modulation on the digital sine sequence to obtain a stepped analog signal of the instrument.

[0055] The filter characteristic reconstruction module is used to reconstruct the filter characteristics of the filter control parameters to obtain the current filter passband that matches the target frequency.

[0056] The harmonic and image filtering module is used to filter out higher harmonics and image frequency components in the stepped analog signal based on the current filter passband, so as to obtain a clean analog signal from the instrument.

[0057] Compared with the prior art, the present invention has the following beneficial effects:

[0058] 1. This invention achieves precise conversion of phase accumulation step size by parsing the target frequency command format, combining the stability verification of the reference clock frequency and bit width alignment; then, it generates high-precision phase sequence and digital sine sequence by triggering phase iteration superposition with clock edge and combining it with coordinate rotation vector iteration algorithm, ensuring the integrity and accuracy of signal generation and laying a reliable signal foundation for the frequency conversion process.

[0059] 2. This invention dynamically adapts filter control parameters through spectral feature mapping, accurately calculates the cutoff frequency using optimization formulas, and achieves optimal fitting between the passband and the target frequency through filter characteristic reconstruction. Furthermore, it significantly improves signal purity through targeted high-order harmonic filtering, precise suppression of image frequencies, and optimization of total harmonic distortion, while enhancing the continuity of the frequency conversion process and significantly improving the frequency conversion efficiency and overall performance of instruments based on direct digital frequency synthesis. Attached Figure Description

[0060] Figure 1 A flowchart illustrating an instrument frequency conversion method based on direct digital frequency synthesis, provided in an embodiment of the present invention;

[0061] Figure 2 A functional block diagram of an instrument frequency conversion system based on direct digital frequency synthesis is provided in an embodiment of the present invention;

[0062] The realization of the objective, functional features and advantages of the present invention will be further explained in conjunction with the embodiments and with reference to the accompanying drawings. Detailed Implementation

[0063] It should be understood that the specific embodiments described herein are merely illustrative of the invention and are not intended to limit the invention.

[0064] This application provides an instrument frequency conversion method based on direct digital frequency synthesis. The execution entity of this instrument frequency conversion method includes, but is not limited to, at least one of the following electronic devices that can be configured to execute the method provided in this application: a server, a terminal, etc. In other words, the instrument frequency conversion method based on direct digital frequency synthesis can be executed by software or hardware installed on a terminal device or a server device. The server includes, but is not limited to, a single server, a server cluster, a cloud server, or a cloud server cluster. The server can be an independent server or a cloud server that provides basic cloud computing services such as cloud services, cloud databases, cloud computing, cloud functions, cloud storage, network services, cloud communication, middleware services, domain name services, security services, content delivery networks (CDNs), and big data and artificial intelligence platforms.

[0065] Reference Figure 1 The diagram shown is a flowchart illustrating an instrument frequency conversion method based on direct digital frequency synthesis according to an embodiment of the present invention. In this embodiment, the instrument frequency conversion method based on direct digital frequency synthesis includes:

[0066] S1. Receive the target frequency command and reference clock signal from the instrument and equipment, and perform frequency tuning word conversion on the target frequency command according to the reference clock frequency in the reference clock signal to obtain the phase accumulation step size of the instrument and equipment.

[0067] In this embodiment of the invention, the step of receiving the target frequency command and reference clock signal from the receiving instrument, and performing frequency tuning word conversion on the target frequency command according to the reference clock frequency in the reference clock signal to obtain the phase accumulation step size of the instrument, includes:

[0068] Receive the target frequency command and reference clock signal from the instrument and equipment, and parse the command format of the target frequency command to obtain the standardized frequency control word of the instrument and equipment;

[0069] The frequency stability of the reference clock frequency in the reference clock signal is verified to obtain the verified reference clock frequency.

[0070] The frequency ratio of the standardized frequency control word to the verified reference clock frequency is calculated to obtain the frequency tuning ratio of the instrument.

[0071] The frequency tuning ratio is bit-width aligned to obtain the phase accumulation step size of the instrument.

[0072] The system receives the target frequency command and reference clock signal sent by the instrument. The target frequency command is a specific signal preset by the instrument to indicate the frequency conversion target, and the reference clock signal is a signal that provides a time reference for the frequency conversion process. According to the preset command format specification of the instrument, the system identifies the key information in the target frequency command that represents frequency control one by one, removes redundant content in the command that is not related to frequency control, and organizes the extracted key information according to a unified character arrangement rule and data representation format. Finally, a standardized frequency control word whose structure and content fully meet the preset requirements of subsequent processing is formed.

[0073] The system continuously acquires the actual output value of the reference clock frequency from the reference clock signal within a preset time period. The preset time period is a fixed duration set to ensure the representativeness of the acquired data. This duration is determined to be 10 milliseconds based on the minimum response period of the instrument's frequency conversion. Each acquired actual output value is compared with a preset reference clock standard value. The preset reference clock standard value is a fixed frequency value calibrated at the factory of the instrument. The preset frequency stability allowable range is ±0.001 Hz of the reference clock standard value. The system determines whether each actual output value is within this allowable range. If all actual output values ​​acquired within 10 milliseconds are within the allowable range, the preset reference clock standard value is determined as the verified reference clock frequency. If there are actual output values ​​that exceed the allowable range, the system re-acquires the actual output value of the reference clock frequency within 10 milliseconds and compares it again until all acquired actual output values ​​meet the allowable range requirements. Then, the preset reference clock standard value is used as the verified reference clock frequency.

[0074] Both the standardized frequency control word and the verified reference clock frequency are presented in specific numerical form. The value corresponding to the standardized frequency control word is used as the dividend, and the value corresponding to the verified reference clock frequency is used as the divisor. A division operation is performed, and the calculation is carried out in sequence according to the decimal numerical operation rules. All valid digits generated in the calculation steps are retained, and no form of numerical rounding is performed. The result that can completely reflect the numerical ratio between the two is obtained directly. This result is the frequency tuning ratio.

[0075] According to the hardware design specifications of the phase accumulator in the instrument, a fixed bit width standard is preset. This standard is the fixed number of bits that the phase accumulator can stably process, specifically set to 32 bits. The frequency tuning ratio is converted into a binary value. If the number of bits of the binary value is less than 32 bits, zeros are continuously padded to the high bits of the binary value until the number of bits reaches exactly 32 bits. If the number of bits of the binary value is more than 32 bits, the lower 32 bits are truncated from the highest bit of the binary value. The 32-bit binary value obtained after the above adjustment is the phase accumulation step size.

[0076] The beneficial effects are that by using a clear signal reception method, fixed instruction parsing rules, quantifiable frequency stability verification standards, standardized ratio calculation process, and determined bit width adjustment method, the obtained phase accumulation step size is ensured to be accurate and stable, fully adapting to the needs of subsequent phase sequence generation of instruments and equipment, providing reliable basic parameter support for the entire instrument frequency conversion process, and ensuring the stability and accuracy of the frequency conversion process.

[0077] S2. Perform spectral feature mapping on the spectral relationship between the target frequency and the reference clock frequency in the target frequency command to obtain the filtering control parameters of the instrument.

[0078] In this embodiment of the invention, the step of mapping the spectral relationship between the target frequency in the target frequency command and the reference clock frequency to obtain the filtering control parameters of the instrument includes:

[0079] Spectral analysis is performed on the target frequency and the reference clock frequency in the target frequency command to obtain the spectral relationship characteristics of the instrument.

[0080] Based on the spectral relationship characteristics, identify the harmonic suppression requirements of the target frequency at the reference clock frequency;

[0081] The harmonic suppression requirement characteristics are mapped to a preset filter type library to obtain the target filter type of the instrument.

[0082] By matching the target filter type with the harmonic suppression requirement characteristics through parameter derivation, the filter parameter range of the instrument is obtained.

[0083] The filter parameter range is optimized and constrained to obtain the filter control parameters of the instrument.

[0084] The formula for calculating the cutoff frequency in the filter control parameters is as follows:

[0085] ;

[0086] In the formula, The cutoff frequency, The target frequency, This is the preset passband reference scaling factor. The preset harmonic suppression dynamic adjustment factor, The reference clock frequency is, The highest harmonic order in the harmonic suppression requirement characteristics. It is the natural logarithm function.

[0087] The target frequency signal in the target frequency command and the reference clock frequency signal after stability verification are acquired. The two signals are synchronously input into the spectrum analysis device. The analysis time is set to 10 times the single frequency conversion cycle of the instrument to ensure that the complete signal cycle is covered. The amplitude, frequency and phase information of the two signals are separated by signal decomposition. The numerical ratio of the target frequency and the reference clock frequency and the interval between the two on the frequency axis are calculated. At the same time, the synchronization relationship of the amplitude of the two signals over time is recorded. These information, including frequency ratio, frequency interval and amplitude synchronization relationship, are integrated to form the spectrum relationship characteristics of the instrument.

[0088] Based on the obtained spectral relationship characteristics, the fundamental frequency amplitude and the amplitudes of each harmonic that is an integer multiple of the fundamental frequency are extracted from the target frequency. The judgment threshold is set to 5% of the fundamental frequency amplitude. The amplitude of each harmonic is compared with the threshold one by one. When the amplitude of a certain harmonic exceeds the threshold, the harmonic is marked as a harmonic that needs to be suppressed. At the same time, the frequency value, amplitude, and multiple relationship with the fundamental frequency of the harmonic are recorded. In addition, the total number of harmonics that need to be suppressed and the frequency interval between each harmonic are counted. These marked and recorded information together constitute the harmonic suppression requirement characteristics of the target frequency at the reference clock frequency.

[0089] The preset filter type library stores four filter types: low-pass, high-pass, band-pass, and band-stop. Each filter type has pre-defined harmonic suppression adaptation conditions. The adaptation condition for a low-pass filter is that the harmonic frequencies to be suppressed are all higher than the target frequency. The adaptation condition for a high-pass filter is that the harmonic frequencies to be suppressed are all lower than the target frequency. The adaptation condition for a band-pass filter is that only the target frequency needs to be retained while suppressing harmonics in all other frequency bands. The adaptation condition for a band-stop filter is that a specific single frequency harmonic needs to be suppressed. The positional relationship between the harmonic frequencies to be suppressed and the target frequency in the harmonic suppression requirement characteristics is compared one by one with the adaptation conditions of each filter in the filter type library. The filter type whose adaptation conditions completely match the harmonic suppression requirement characteristics is selected as the target filter type for the instrument.

[0090] For a given target filter type, based on information such as harmonic frequency, amplitude, and number of harmonics to be suppressed, parameter derivation rules are formulated. If the target filter type is a low-pass filter, the lowest boundary of the cutoff frequency is determined based on the highest harmonic frequency to be suppressed, which is 0.8 times the highest harmonic frequency. The highest boundary of the cutoff frequency is determined based on the target frequency, which is 1.2 times the target frequency. Then, the minimum value of the filter order is determined based on the ratio of the harmonic amplitude to the fundamental frequency amplitude to be suppressed. This minimum value must satisfy the requirement that the harmonic amplitude attenuates to less than 1% of the fundamental frequency amplitude. The maximum value of the order is set to 20 to avoid excessively increasing the computational load on the equipment. The cutoff frequency range and order range obtained through the above derivation together constitute the filter parameter range of the instrument equipment.

[0091] The cutoff frequency in the filter control parameters is calculated through specific logic. The key information required all originates from the explicit output of the preceding process. The target frequency is the core frequency value in the standardized frequency control word that characterizes the instrument's frequency conversion target. The passband reference scaling factor is a fixed value determined by repeatedly verifying and analyzing the minimum ratio requirement of the filter passband covering the target frequency in each scenario based on a large amount of filter passband adaptation test data from different combinations of target frequency and reference clock frequency. The harmonic suppression dynamic adjustment factor is a fixed coefficient determined by repeatedly testing and adjusting the harmonic suppression effect under different highest harmonic orders based on the verification data. This adjustment is made to address the impact of different harmonic orders on the filter passband. The reference clock frequency is the verified reference clock frequency after a preset duration stability check, where all actual output values ​​are within the allowable range. The highest harmonic order is the highest order of the harmonic to be suppressed in the harmonic suppression requirement characteristics. In the calculation, the ratio of the target frequency to the reference clock frequency is first obtained and multiplied by the highest harmonic order. After adding 1, the natural logarithm is processed. The result is multiplied by the harmonic suppression dynamic adjustment factor, then added to the passband reference ratio factor, and finally multiplied by the target frequency to obtain the cutoff frequency that accurately matches the current harmonic suppression requirement. This calculation logic achieves accurate adaptation of the cutoff frequency by integrating key information reflecting the current spectrum relationship and harmonic suppression requirements.

[0092] The optimization constraints are set as follows: the filter's insertion loss should not exceed 3 dB, and the transition band width should not exceed 0.1 times the target frequency. Within the filter parameter range, each possible parameter combination is verified by iterating from the middle value of the parameter range towards both ends. First, the combination of the middle value of the cutoff frequency and order is verified. It is checked whether the filter corresponding to this combination meets the constraints of insertion loss and transition band width. If it meets the constraints, the adjacent parameter combinations are verified. If it does not meet the constraints, the parameter combination is excluded. This process is continued until all parameter combinations are covered. From all parameter combinations that meet the constraints, the parameter combination with the smallest transition band width and the smallest insertion loss is selected. This parameter combination is the filter control parameter of the instrument.

[0093] The beneficial effects are that by using a clear spectrum analysis method, a quantitative harmonic identification standard, a precise filter type matching rule, a standardized parameter derivation process, a cutoff frequency calculation logic based on key information of the actual scenario, and a strict optimization constraint solution method, the obtained filter control parameters are ensured to be fully compatible with the harmonic suppression requirements of the target frequency. This provides accurate and reliable parameter support for subsequent filter characteristic reconstruction, ensures that the filtering process can specifically suppress harmonic components, and improve the purity of the signal after the instrument is frequency converted.

[0094] S3. Based on the phase accumulation step size and the reference clock signal, generate the phase sequence of the instrument and equipment, and perform coordinate rotation vector iteration on the current phase value in the phase sequence to obtain the digital sine sequence of the instrument and equipment;

[0095] In this embodiment of the invention, the step of generating a phase sequence of the instrument based on the phase accumulation step size and the reference clock signal, and performing coordinate rotation vector iteration on the current phase value in the phase sequence to obtain a digital sine sequence of the instrument, includes:

[0096] The phase value of the phase accumulator in the instrument is initialized to zero, and triggered by the clock edge of the reference clock signal, the phase accumulation step size is iteratively superimposed with the real-time phase value of the phase accumulator to obtain the phase sequence of the instrument.

[0097] Based on the bit width of the phase accumulation step size, the current phase value in the phase sequence is truncated to obtain the phase angle to be iterated for the current phase value;

[0098] Based on a preset initial rotation vector, the phase angle to be iterated is successively rotated to approximate it, and the convergent rotation coordinates of the phase angle to be iterated are obtained.

[0099] The convergent rotating coordinates are orthogonally decomposed and extracted to obtain the digital sine sequence of the instrument.

[0100] The initial phase value of the phase accumulator in the instrument is set to zero. This zero value serves as the reference starting point for phase measurement. The trigger edge of the reference clock signal is set to the rising edge. When the level of the reference clock signal jumps from low to high, the phase accumulation operation is triggered. The pre-obtained phase accumulation step size is added to the current real-time phase value in the phase accumulator to obtain a new real-time phase value. This addition operation is repeated every time the rising edge of the reference clock signal arrives. After triggering multiple times, all the real-time phase values ​​generated in chronological order are collected and set together to form the phase sequence of the instrument.

[0101] The bit width of the phase accumulation step is 32 bits, which is determined after the previous bit width alignment. Using this 32-bit bit width as the standard, the number of bits of each current phase value in the phase sequence is truncated, retaining only the lower 32 bits of the current phase value and discarding the higher bits. The truncated 32 bits are then mapped to an angle range of 0 to 2π through a linear transformation. The transformation standard is that the maximum value of the 32 bits corresponds to 2π and the minimum value corresponds to 0. The angle value obtained after this truncation and mapping process is the phase angle to be iterated for the current phase value.

[0102] The preset initial rotation vector is the (1,0) vector in a two-dimensional rectangular coordinate system, with the direction of the positive x-axis. The phase angle to be iterated is divided into 8 equal sub-angles. During the first rotation, the initial rotation vector is rotated counterclockwise according to the first sub-angle to obtain the vector after the first rotation. The angle between this vector and the target vector corresponding to the phase angle to be iterated is calculated. If the angle is greater than 0.001 radians, the rotation continues in the same direction according to the second sub-angle. If the angle is less than 0.001 radians, the same sub-angle is rotated clockwise. This rotation and angle judgment operation is repeated until the angle between the rotated vector and the target vector is less than or equal to 0.001 radians. At this time, the coordinates of the rotation vector in the two-dimensional rectangular coordinate system are the convergent rotation coordinates of the phase angle to be iterated.

[0103] An orthogonal decomposition method is used to process the convergent rotation coordinates, decomposing them into x-axis and y-axis components. The x-axis component corresponds to the cosine value, and the y-axis component corresponds to the sine value. The y-axis component value is extracted from each convergent rotation coordinate, and each y-axis component value is converted into a 16-bit binary number. During the conversion, the maximum value of the 16-bit binary number corresponding to the maximum value of the y-axis component and the minimum value of the 16-bit binary number corresponding to the minimum value of the y-axis component are used as the conversion standard. These 16-bit binary numbers extracted and converted in chronological order are arranged in order to form the digital sine sequence of the instrument.

[0104] The beneficial effects are that, through clear phase initialization standards, fixed clock triggering methods, standardized phase interception and mapping rules, quantifiable rotation approximation convergence conditions, and clear orthogonal decomposition and digital conversion processes, the generated digital sine sequence has accurate phase and amplitude characteristics and strong timing consistency, providing a high-quality digital signal foundation for subsequent pulse amplitude modulation stages and ensuring the stability and accuracy of signal conversion during the instrument's frequency conversion process.

[0105] S4. Pulse amplitude modulation is performed on the digital sine sequence to obtain the stepped analog signal of the instrument.

[0106] In this embodiment of the invention, the step of pulse amplitude modulation of the digital sine sequence to obtain the stepped analog signal of the instrument includes:

[0107] The digital sample values ​​in the digital sine sequence are converted from digital to analog to obtain the analog level values ​​of the digital sample values;

[0108] Based on the timing of the reference clock signal, the analog level value is time-driven to obtain a constant amplitude pulse of the analog level value;

[0109] The constant amplitude pulse is reconstructed to obtain the pulse amplitude modulation signal of the constant amplitude pulse;

[0110] The pulse amplitude modulation signal is time-domain waveform preserved to obtain the stepped analog signal of the instrument.

[0111] Each digital sample value in the digital sine sequence is a 16-bit binary digital quantity, with a value range of 0 to 65535. The preset output range of the analog level value is 0 to 5 volts. A linear mapping relationship between the digital sample value and the analog level value is established. The digital sample value 0 corresponds to a 0-volt analog level, and the digital sample value 65535 corresponds to a 5-volt analog level. For each digital sample value between 0 and 65535, the corresponding analog level value is calculated in the same proportion. During the calculation, the digital sample value is divided by 65535 and then multiplied by 5 volts. The result is the analog level value corresponding to that digital sample value.

[0112] The period of the reference clock signal is a fixed value, which is determined by the reference clock frequency. The trigger edge of the preset reference clock signal is the rising edge. When the rising edge arrives, the analog level value at the current moment is extracted and used as the amplitude of the pulse to generate a rectangular pulse. The width of the pulse is equal to the period of the reference clock signal, and the amplitude of the pulse remains unchanged as the extracted analog level value. Each analog level value corresponds to the generation of such a rectangular pulse. These pulses are output sequentially according to the timing of the reference clock signal to obtain a constant amplitude pulse.

[0113] All constant amplitude pulses are arranged in an orderly manner according to the time sequence of the digital sample values ​​in the digital sine sequence. During the arrangement, it is ensured that the end time of the previous pulse is completely connected with the start time of the next pulse, and there is no overlap or gap between the pulses. The amplitude of each constant amplitude pulse remains unchanged according to its corresponding analog level value, and the pulse width is always the period of the reference clock signal. The continuous pulse sequence formed by this orderly arrangement is the pulse amplitude modulation signal of the constant amplitude pulse.

[0114] A dedicated waveform holding circuit is used to process the pulse amplitude modulation signal. When each constant amplitude pulse is input, the circuit locks the amplitude of the pulse and outputs it continuously. The locking time is equal to the period of the reference clock signal. During the locking time, the output amplitude remains unchanged regardless of whether the subsequent signal changes, until the next constant amplitude pulse is input, at which point the circuit unlocks and locks a new pulse amplitude. Through this holding method, the continuously output signal forms a stepped waveform. The height of each step is the amplitude of the corresponding constant amplitude pulse, and the width of each step is equal to the period of the reference clock signal, ultimately resulting in a stepped analog signal for the instrument.

[0115] The beneficial effects are that by establishing clear digital-to-analog mapping standards, fixed timing drive rules, standardized signal recombination methods, and stable waveform holding mechanisms, the generated stepped analog signals are ensured to have accurate timing and stable amplitude, accurately reproduce the characteristics of digital sine sequences, provide a high-quality analog signal foundation for subsequent filtering stages, and effectively guarantee the integrity and reliability of signal conversion during the instrument's frequency conversion process.

[0116] S5. Reconstruct the filtering characteristics of the filtering control parameters to obtain the current filtering passband that matches the target frequency;

[0117] In this embodiment of the invention, the step of reconstructing the filter characteristics of the filter control parameters to obtain a current filter passband that matches the target frequency includes:

[0118] Extract the cutoff frequency and filter order from the filter control parameters to generate the filter structure parameters of the instrument.

[0119] Based on the filter structure parameters, the amplitude response and phase response of the filter in the instrument are parametrically reconstructed to obtain the reconstructed frequency response characteristics of the instrument.

[0120] Based on the target frequency and the reconstructed frequency response characteristics, the passband boundary and transition band slope of the filter are optimally fitted to obtain the current filter passband that matches the target frequency.

[0121] Core values ​​are selected from the set of filter control parameters obtained through previous optimization and constraint solving. All parameters in this set follow a unified identification rule, with each parameter corresponding to a unique name label. The labels "cutoff frequency" and "filter order" are preset fixed identifiers. By identifying these fixed identifiers, the corresponding two specific values ​​are accurately extracted. These two values ​​are accurate to six decimal places, directly quantifying the core control indicators of the filter. The target filter type is a clearly defined type obtained by adapting the harmonic suppression requirements to a preset filter type library. The extracted cutoff frequency, filter order, and target filter type are associated and integrated, together constituting the filter structure parameters of the instrument. The target filter type clarifies whether the filter operates in low-pass, high-pass, band-pass, or band-stop mode; the cutoff frequency defines the critical frequency boundary that the filter allows or blocks signals; and the filter order determines the attenuation level of the filter for signals exceeding the cutoff frequency.

[0122] The filter structure parameters include the target filter type, cutoff frequency, and filter order. For the four preset filter types of low-pass, high-pass, band-pass, and band-stop, there are unique and fixed amplitude response and phase response reconstruction rules. The amplitude response rules for low-pass filters are as follows: when the frequency is below the cutoff frequency, the amplitude remains at 95% of the input signal amplitude; when the frequency is above the cutoff frequency, the amplitude decays at a rate corresponding to the filter order. For order 1, the decay rate is 20 dB per decade; for each additional order, the decay rate increases by 20 dB per decade. The amplitude response rules for high-pass filters are as follows: when the frequency is above the cutoff frequency, the amplitude remains at 95% of the input signal amplitude; when the frequency is below the cutoff frequency, it decays according to the same order decay rule as the low-pass filter. The amplitude response rules for band-pass filters are as follows: when the frequency is within the passband defined by the cutoff frequency, the amplitude remains at 95%; outside the passband, it decays according to the corresponding order decay rule. The amplitude response rules for band-stop filters are as follows: at the target suppression frequency and within 0.5% of the cutoff frequency before and after it, the amplitude decays to below 30% of the input signal; in other frequency bands, it maintains 95% of the amplitude. The phase response rules for the four types of filters are unified as follows: the phase offset corresponding to each frequency in the passband is controlled within ±5 degrees; the phase offset in the transition band and stopband changes according to a linear law; and the phase offset difference between adjacent frequency points does not exceed 3 degrees. Based on the above rules for corresponding types, combined with the specific values ​​in the filter structure parameters, using 1 / 1000 of the reference clock frequency as the frequency step size, the amplitude and phase values ​​corresponding to all frequency points within the range from 0 to twice the reference clock frequency are calculated point by point. When calculating the amplitude value, the specific value is derived according to the attenuation rule and the order. The phase value is calculated according to the linear change rule to calculate the offset of adjacent frequency points. All frequency points and their corresponding amplitude and phase values ​​are arranged in ascending order of frequency to form the reconstructed frequency response characteristics of the instrument.

[0123] The target frequency is the core signal frequency that the instrument's frequency conversion must retain. It must be completely contained within the current filter passband. Based on the reconstructed frequency response characteristics, the criteria for determining the passband boundary are set as follows: the amplitude response must not be less than 95% of the input signal amplitude. The criteria for determining the transition band slope are: the ratio of the frequency range where the amplitude decays from 95% to 30% within the transition band to the cutoff frequency must not exceed 10%. Using 1 / 1000 of the reference clock frequency as the traversal step size, with the target frequency as the center reference point, the amplitude value in the reconstructed frequency response characteristics is checked point by point to the left of the frequency axis. When the first frequency point with an amplitude value lower than 95% is encountered, the previous frequency point is determined as the left passband boundary. Similarly, the same step size is used to check point by point to the right of the frequency axis. When the first frequency point with an amplitude value lower than 95% is encountered, the previous frequency point is determined as the right passband boundary. Next, the transition band width is calculated, which is the sum of the widths of the right and left transition bands. If the ratio of this width to the cutoff frequency exceeds 10%, the filter order is increased by 1, and the amplitude response and transition band width are recalculated. This adjustment process is repeated until the ratio of the transition band width to the cutoff frequency does not exceed 10%. The final determined passband boundary and the adjusted transition band slope together constitute the current filter passband that matches the target frequency. It is also necessary to verify that the target frequency is between the left and right passband boundaries to ensure that the target frequency is completely contained.

[0124] The beneficial effects are that by using clear parameter identification rules, precise core value extraction methods, fixed response reconstruction rules covering all filter types, unified and specific frequency step size settings, and quantifiable passband boundary and transition band slope judgment criteria, the reconstructed frequency response characteristics are ensured to be accurate and reliable. The final current filter passband can achieve precise matching with the target frequency, with clear passband boundaries and reasonable transition band slopes. This provides a clear and reliable filtering basis for subsequent targeted filtering of higher harmonics and image frequency components, effectively ensuring the accuracy and effectiveness of the filtering process and avoiding target signal loss or interference signal residue due to improper filter passband settings.

[0125] S6. Based on the current filter passband, filter out the higher harmonics and image frequency components in the stepped analog signal to obtain the pure analog signal of the instrument.

[0126] In this embodiment of the invention, the step of filtering out higher harmonics and image frequency components from the stepped analog signal based on the current filter passband to obtain a clean analog signal for the instrument includes:

[0127] Based on the current passband range and stopband attenuation requirements, configure the center frequency and bandwidth parameters for the filter in the instrument.

[0128] The stepped analog signal is input to a pre-configured filter for bandpass filtering to obtain the preliminary analog signal of the instrument.

[0129] Based on the relative relationship between the target frequency and the reference clock frequency, the preliminary analog signal is subjected to image frequency suppression to obtain the suppressed analog signal of the instrument.

[0130] The total harmonic distortion of the suppressed analog signal is optimized to obtain a clean analog signal from the instrument.

[0131] The step of performing image frequency suppression on the preliminary analog signal based on the relative relationship between the target frequency and the reference clock frequency to obtain the suppressed analog signal of the instrument includes:

[0132] Based on the target frequency and the reference clock frequency, the spectral characteristics of the stepped analog signal are mapped and identified to obtain the underlying coordinates of the potential mirror frequency of the stepped analog signal.

[0133] Based on the frequency multiplication relationship of the reference clock frequency, the underlying coordinates of the potential mirror frequency are multiplied by frequency to obtain the specific spectral location set of the underlying coordinates of the potential mirror frequency.

[0134] Based on the specific spectral location set, the configuration parameters of the image frequency suppression filter in the instrument are generated in a regularized manner to obtain the stopband center frequency and stopband width parameters of the image frequency suppression filter.

[0135] By applying the stopband center frequency and the stopband width parameters, the preliminary analog signal is band-stop filtered to obtain the suppressed analog signal of the instrument.

[0136] The left and right passband boundaries of the current filter are specific frequency values ​​determined after reconstructing the filter characteristics. The center frequency is set as the average of these two boundary values. The calculation involves first performing an exact addition operation between the left and right passband boundary values, retaining six decimal places, and then dividing the result by 2. The final value, retaining six decimal places, is the filter's center frequency. The bandwidth parameter is the difference between the right and left passband boundary values, obtained through direct subtraction, also retaining six decimal places. The stopband attenuation requirement is set to attenuate the amplitude to below 30% of the input signal. This threshold is determined based on industry-standard standards for the purity of filtered signals, ensuring that the attenuated harmonic signals do not interfere with the target signal. Based on this attenuation requirement and the calculated center frequency and bandwidth parameters, the active RC filter structure used in the instrument is configured, specifying the exact resistor and capacitor selections to complete the filter's hardware parameter adaptation.

[0137] The previously generated stepped analog signal is input to a pre-configured filter via a shielded coaxial transmission line. This transmission line effectively reduces the impact of external electromagnetic interference on the signal. The filter employs an active RC filter structure. Its internal circuitry sets the resistance and capacitance values ​​according to the configured center frequency and bandwidth parameters, allowing only signals within the passband defined by the center frequency and bandwidth to pass through. The amplitude attenuation of the signal within the passband is controlled within 5%. When high-order harmonic signals outside the passband pass through the filter, their amplitude is stably attenuated to below 30% of the input signal through the superposition attenuation effect of multiple RC circuits. After the filter's screening and attenuation processing, the output signal, which retains the target frequency components, initially filters out high-order harmonics, and has an amplitude fluctuation of no more than 3%, is the preliminary analog signal of the instrument.

[0138] The target frequency and the reference clock frequency after stability verification are obtained. A stepped analog signal is input into a spectrum analyzer with a sampling rate of 10 times the reference clock frequency. The analysis step size is set to 1 / 1000 of the reference clock frequency. The spectrum characteristics from 0 to twice the reference clock frequency are analyzed, and the amplitude and phase information of each frequency point are extracted point by point. The difference between the reference clock frequency and the target frequency is calculated. The spectral coordinates corresponding to this difference are the basic coordinates of the potential mirror frequency. Simultaneously, the amplitude comparison module built into the spectrum analyzer verifies whether the amplitude corresponding to this coordinate exceeds 10% of the target frequency amplitude. This threshold is derived from experiments on the critical value of mirror frequency interference to the target signal. If it exceeds this threshold, the coordinate is confirmed as a valid basic coordinate of the potential mirror frequency; if it does not exceed this threshold, the signal corresponding to this coordinate is determined to have no substantial interference to the target signal and is not included in the potential mirror frequency range.

[0139] The preset multiplication factor range of the reference clock frequency is 1 to 5 times. This range was determined based on experimental data. Beyond 5 times, the amplitude of the image frequency will be less than 1% of the target frequency, and its impact on the signal is negligible, so it does not need to be included in the suppression range. Based on this multiplication factor relationship, the basic coordinates of the potential image frequency are extended and calculated. The differences between the 1st, 2nd, 3rd, 4th, and 5th times reference clock frequencies and the target frequency are calculated sequentially. In the calculation process, six significant decimal places are retained. All frequency coordinates obtained through multiplication factor calculation are collected and organized in ascending order of value to form a specific spectral location set of the basic coordinates of the potential image frequency. Each coordinate is labeled with the corresponding multiplication factor.

[0140] For each frequency coordinate in a specific spectral location set, it is directly set as the stopband center frequency of the image frequency suppression filter, retaining six decimal places. The stopband width parameter is uniformly set to 5% of the target frequency value. This width is determined experimentally based on the spectral diffusion characteristics of the image frequency, ensuring complete coverage of the image frequency and surrounding interference bands at the corresponding spectral location. During calculation, the target frequency value is multiplied by 5%, and the specific width value is obtained by retaining six decimal places. The image frequency suppression filter adopts a digital band-stop filter structure. Following the rule of pairing each stopband center frequency with a corresponding stopband width, all configuration parameters of the filter are generated sequentially, including digital filter coefficients, delay time, etc., ultimately resulting in a set of stopband center frequency and stopband width parameters. The parameter set is arranged in the order of the spectral location set for easy sequential loading and execution of the filter.

[0141] The initial analog signal is input to a pre-configured image frequency suppression filter via a high-speed data transmission interface. This filter is implemented in real-time using FPGA hardware, with a filtering delay not exceeding one cycle of the reference clock signal. The filter attenuates the image frequency signal within the stopband range, ensuring that the attenuated amplitude does not exceed 30% of the input signal through internal digital logic processing. Simultaneously, it compensates for the amplitude of signals outside the stopband range, maintaining amplitude fluctuations below 2%. After this band-stop filtering process, the output signal is the suppressed analog signal for the instrument. Before output, the attenuation effect is verified in real-time by an amplitude detection circuit to ensure it meets requirements.

[0142] A high-precision harmonic distortion (THD) detection device is used to detect the total harmonic distortion (THD) of the suppressed analog signal. The device has a detection accuracy of 0.01%, and the allowable THD threshold is set to 1%, which conforms to the output standard of high-precision instrument frequency converter signals. During detection, the fundamental frequency component and each harmonic component in the analog signal are first separated using the device's built-in signal separation module. Then, the amplitude of the fundamental frequency and the total amplitude of all harmonic components are calculated separately. The ratio of these two values ​​is the THD value, and the calculation is rounded to three decimal places. If the THD result shows that the THD exceeds 1%, the stopband attenuation parameter of the image frequency suppression filter is adjusted through the filter parameter adjustment interface. Each adjustment increases the attenuation intensity by 5%. After adjustment, the signal is re-input into the harmonic distortion detection device for detection. This adjustment and detection process is repeated, up to a maximum of 5 times. If the requirement is still not met, the process returns to the spectrum analysis step to re-identify the image frequency until the THD detection result is ≤1%. At this point, the output signal is the pure analog signal of the instrument. The detection results and the final filter parameters are recorded simultaneously during output for easy traceability and maintenance.

[0143] The beneficial effects are as follows: through clear filter parameter calculation rules, specific hardware selection and configuration standards, high-precision spectrum analysis and image frequency identification process, octave range and stopband width settings based on experimental data, quantifiable attenuation and distortion detection thresholds, and a complete parameter adjustment and verification mechanism, it is ensured that high-order harmonics and image frequency components are accurately and thoroughly filtered out. The resulting pure analog signal has a total harmonic distortion that strictly meets the preset requirements, high signal amplitude stability, and small phase shift, effectively improving the frequency conversion quality and signal output reliability of the instrument based on direct digital frequency synthesis. At the same time, the operation steps of the entire process are clear, the parameter settings are based on evidence, and the results are verifiable, ensuring the reproducibility of the technical solution.

[0144] like Figure 2 The diagram shown is a functional block diagram of an instrument frequency conversion system based on direct digital frequency synthesis provided in an embodiment of the present invention.

[0145] The instrument frequency conversion system 100 based on direct digital frequency synthesis described in this invention can be installed in an electronic device. Depending on the functions implemented, the instrument frequency conversion system 100 may include a frequency tuning word conversion module 101, a spectrum feature mapping module 102, a digital sine sequence generation module 103, a pulse amplitude modulation module 104, a filter characteristic reconstruction module 105, and a harmonic and image filtering module 106. The module described in this invention can also be called a unit, referring to a series of computer program segments that can be executed by the processor of an electronic device and perform a fixed function, stored in the memory of the electronic device.

[0146] In this embodiment, the functions of each module / unit are as follows:

[0147] The frequency tuning word conversion module 101 is used to receive the target frequency command and reference clock signal of the instrument and equipment, and to convert the target frequency command into a frequency tuning word according to the reference clock frequency in the reference clock signal, so as to obtain the phase accumulation step size of the instrument and equipment.

[0148] The spectrum feature mapping module 102 is used to perform spectrum feature mapping on the spectrum relationship between the target frequency and the reference clock frequency in the target frequency command to obtain the filtering control parameters of the instrument.

[0149] The digital sine sequence generation module 103 is used to generate the phase sequence of the instrument based on the phase accumulation step size and the reference clock signal, and to perform coordinate rotation vector iteration on the current phase value in the phase sequence to obtain the digital sine sequence of the instrument.

[0150] The pulse amplitude modulation module 104 is used to perform pulse amplitude modulation on the digital sine sequence to obtain the step-shaped analog signal of the instrument.

[0151] The filter characteristic reconstruction module 105 is used to reconstruct the filter characteristics of the filter control parameters to obtain the current filter passband that matches the target frequency.

[0152] The harmonic and image filtering module 106 is used to filter out the higher harmonics and image frequency components in the stepped analog signal according to the current filter passband, so as to obtain the pure analog signal of the instrument.

[0153] In the several embodiments provided by this invention, it should be understood that the disclosed methods and systems can be implemented in other ways. For example, the system embodiments described above are merely illustrative; for instance, the division of modules is only a logical functional division, and other division methods may be used in actual implementation.

[0154] The modules described as separate components may or may not be physically separate. The components shown as modules may or may not be physical units; that is, they may be located in one place or distributed across multiple network units. Some or all of the modules can be selected to achieve the purpose of this embodiment according to actual needs.

[0155] Furthermore, the functional modules in the various embodiments of the present invention can be integrated into one processing unit, or each unit can exist physically separately, or two or more units can be integrated into one unit. The integrated unit can be implemented in hardware or in the form of hardware plus software functional modules.

[0156] It will be apparent to those skilled in the art that the present invention is not limited to the details of the exemplary embodiments described above, and that the present invention can be implemented in other specific forms without departing from the spirit or essential characteristics of the present invention.

[0157] This application embodiment can acquire and process relevant data based on artificial intelligence technology. Artificial intelligence is the theory, method, technology, and application system that uses digital computers or machines controlled by digital computers to simulate, extend, and expand human intelligence, perceive the environment, acquire knowledge, and use that knowledge to obtain optimal results.

[0158] Finally, it should be noted that the above embodiments are only used to illustrate the technical solutions of the present invention and are not intended to limit it. Although the present invention has been described in detail with reference to preferred embodiments, those skilled in the art should understand that modifications or equivalent substitutions can be made to the technical solutions of the present invention without departing from the spirit and scope of the technical solutions of the present invention.

Claims

1. An instrument frequency conversion method based on direct digital frequency synthesis, characterized in that, The method includes: S1. Receive the target frequency command and reference clock signal from the instrument and equipment, and perform frequency tuning word conversion on the target frequency command according to the reference clock frequency in the reference clock signal to obtain the phase accumulation step size of the instrument and equipment. S2. Perform spectral feature mapping on the spectral relationship between the target frequency and the reference clock frequency in the target frequency command to obtain the filtering control parameters of the instrument. S3. Based on the phase accumulation step size and the reference clock signal, generate the phase sequence of the instrument and equipment, and perform coordinate rotation vector iteration on the current phase value in the phase sequence to obtain the digital sine sequence of the instrument and equipment; S4. Pulse amplitude modulation is performed on the digital sine sequence to obtain the stepped analog signal of the instrument. S5. Reconstruct the filtering characteristics of the filtering control parameters to obtain the current filtering passband that matches the target frequency; S6. Based on the current filter passband, filter out the higher harmonics and image frequency components in the stepped analog signal to obtain the pure analog signal of the instrument.

2. The instrument frequency conversion method based on direct digital frequency synthesis as described in claim 1, characterized in that, The receiving instrument receives a target frequency command and a reference clock signal, and performs frequency tuning word conversion on the target frequency command based on the reference clock frequency in the reference clock signal to obtain the phase accumulation step size of the instrument, including: Receive the target frequency command and reference clock signal from the instrument and equipment, and parse the command format of the target frequency command to obtain the standardized frequency control word of the instrument and equipment; The frequency stability of the reference clock frequency in the reference clock signal is verified to obtain the verified reference clock frequency. The frequency ratio of the standardized frequency control word to the verified reference clock frequency is calculated to obtain the frequency tuning ratio of the instrument. The frequency tuning ratio is bit-width aligned to obtain the phase accumulation step size of the instrument.

3. The instrument frequency conversion method based on direct digital frequency synthesis as described in claim 1, characterized in that, The step of mapping the spectral relationship between the target frequency in the target frequency command and the reference clock frequency to obtain the filter control parameters of the instrument includes: Spectral analysis is performed on the target frequency and the reference clock frequency in the target frequency command to obtain the spectral relationship characteristics of the instrument. Based on the spectral relationship characteristics, identify the harmonic suppression requirements of the target frequency at the reference clock frequency; The harmonic suppression requirement characteristics are mapped to a preset filter type library to obtain the target filter type of the instrument. By matching the target filter type with the harmonic suppression requirement characteristics through parameter derivation, the filter parameter range of the instrument is obtained. The filter parameter range is optimized and constrained to obtain the filter control parameters of the instrument.

4. The instrument frequency conversion method based on direct digital frequency synthesis as described in claim 3, characterized in that, The formula for calculating the cutoff frequency in the filter control parameters is as follows: ； In the formula, The cutoff frequency, The target frequency, This is the preset passband reference scaling factor. The preset harmonic suppression dynamic adjustment factor, The reference clock frequency is, The highest harmonic order in the harmonic suppression requirement characteristics. It is the natural logarithm function.

5. The instrument frequency conversion method based on direct digital frequency synthesis as described in claim 1, characterized in that, The step of generating a phase sequence of the instrument based on the phase accumulation step size and the reference clock signal, and performing coordinate rotation vector iteration on the current phase value in the phase sequence to obtain a digital sine sequence of the instrument, includes: The phase value of the phase accumulator in the instrument is initialized to zero, and triggered by the clock edge of the reference clock signal, the phase accumulation step size is iteratively superimposed with the real-time phase value of the phase accumulator to obtain the phase sequence of the instrument. Based on the bit width of the phase accumulation step size, the current phase value in the phase sequence is truncated to obtain the phase angle to be iterated for the current phase value; Based on a preset initial rotation vector, the phase angle to be iterated is successively rotated to approximate it, and the convergent rotation coordinates of the phase angle to be iterated are obtained. The convergent rotating coordinates are orthogonally decomposed and extracted to obtain the digital sine sequence of the instrument.

6. The instrument frequency conversion method based on direct digital frequency synthesis as described in claim 1, characterized in that, The stepwise analog signal of the instrument is obtained by pulse amplitude modulation of the digital sine sequence, including: The digital sample values ​​in the digital sine sequence are converted from digital to analog to obtain the analog level values ​​of the digital sample values; Based on the timing of the reference clock signal, the analog level value is time-driven to obtain a constant amplitude pulse of the analog level value; The constant amplitude pulse is reconstructed to obtain the pulse amplitude modulation signal of the constant amplitude pulse; The pulse amplitude modulation signal is time-domain waveform preserved to obtain the stepped analog signal of the instrument.

7. The instrument frequency conversion method based on direct digital frequency synthesis as described in claim 1, characterized in that, The step of reconstructing the filter characteristics of the filter control parameters to obtain the current filter passband that matches the target frequency includes: Extract the cutoff frequency and filter order from the filter control parameters to generate the filter structure parameters of the instrument. Based on the filter structure parameters, the amplitude response and phase response of the filter in the instrument are parametrically reconstructed to obtain the reconstructed frequency response characteristics of the instrument. Based on the target frequency and the reconstructed frequency response characteristics, the passband boundary and transition band slope of the filter are optimally fitted to obtain the current filter passband that matches the target frequency.

8. The instrument frequency conversion method based on direct digital frequency synthesis as described in claim 1, characterized in that, The step of filtering out higher harmonics and image frequency components from the stepped analog signal based on the current filter passband to obtain a clean analog signal for the instrument includes: Based on the current passband range and stopband attenuation requirements, configure the center frequency and bandwidth parameters for the filter in the instrument. The stepped analog signal is input to a pre-configured filter for bandpass filtering to obtain the preliminary analog signal of the instrument. Based on the relative relationship between the target frequency and the reference clock frequency, the preliminary analog signal is subjected to image frequency suppression to obtain the suppressed analog signal of the instrument. The total harmonic distortion of the suppressed analog signal is optimized to obtain a clean analog signal from the instrument.

9. The instrument frequency conversion method based on direct digital frequency synthesis as described in claim 8, characterized in that, The step of performing image frequency suppression on the preliminary analog signal based on the relative relationship between the target frequency and the reference clock frequency to obtain the suppressed analog signal of the instrument includes: Based on the target frequency and the reference clock frequency, the spectral characteristics of the stepped analog signal are mapped and identified to obtain the underlying coordinates of the potential mirror frequency of the stepped analog signal. Based on the frequency multiplication relationship of the reference clock frequency, the underlying coordinates of the potential mirror frequency are multiplied by frequency to obtain the specific spectral location set of the underlying coordinates of the potential mirror frequency. Based on the specific spectral location set, the configuration parameters of the image frequency suppression filter in the instrument are generated in a regularized manner to obtain the stopband center frequency and stopband width parameters of the image frequency suppression filter. By applying the stopband center frequency and the stopband width parameters, the preliminary analog signal is band-stop filtered to obtain the suppressed analog signal of the instrument.

10. An instrument frequency conversion system based on direct digital frequency synthesis, characterized in that, The system for implementing the instrument frequency conversion method based on direct digital frequency synthesis as described in claim 1 includes: The frequency tuning word conversion module is used to receive the target frequency command and reference clock signal of the instrument and equipment, and to convert the target frequency command into a frequency tuning word according to the reference clock frequency in the reference clock signal, so as to obtain the phase accumulation step size of the instrument and equipment. The spectrum feature mapping module is used to perform spectrum feature mapping on the spectrum relationship between the target frequency and the reference clock frequency in the target frequency command to obtain the filtering control parameters of the instrument. A digital sine sequence generation module is used to generate a phase sequence of the instrument based on the phase accumulation step size and the reference clock signal, and to perform coordinate rotation vector iteration on the current phase value in the phase sequence to obtain a digital sine sequence of the instrument. A pulse amplitude modulation module is used to perform pulse amplitude modulation on the digital sine sequence to obtain a stepped analog signal of the instrument. The filter characteristic reconstruction module is used to reconstruct the filter characteristics of the filter control parameters to obtain the current filter passband that matches the target frequency. The harmonic and image filtering module is used to filter out higher harmonics and image frequency components in the stepped analog signal based on the current filter passband, so as to obtain a clean analog signal from the instrument.

Images