Spurious-Free Dynamic Range Augmentation for Digital-to-Analog Converters using PSO-PID Adaptive Algorithms
Article information
Abstract
In this paper, we propose a new method for enhancing the spurious-free dynamic range (SFDR) of digital-to-analog converters (DACs) by implementing a digital predistorter to suppress harmonic distortion. First, the particle swarm optimization (PSO) algorithm is used to optimize the proportional-integral-derivative (PID) parameters, after which the PSO-PID algorithm is implemented to identify the best signal featuring the least harmonic distortion. Next, the Hilbert transform-based memory polynomial (HTBMP) model is employed to accurately represent the harmonic predistorter for the DAC. Finally, using a single-tone sine signal in shortwave frequency bands, it is demonstrated that the proposed HTBMP predistortion model can enhance the SFDR of a DAC by at least 9 dB.
I. Introduction
In communication systems, the output signal quality of a digital-to-analog converter (DAC) directly impacts the performance of the modulation/demodulation system, thereby influencing both signal clarity and overall system efficiency. In this context, enhancing the spurious-free dynamic range (SFDR) of a DAC enables it to maintain consistent performance across diverse working environments and signal conditions, rendering it suitable for a broader range of applications. For instance, in the case of applications that require substantial signal resolution, such as radar and satellite communication, enhancing the SFDR of the DAC can augment system resolution, thereby bolstering accuracy and reliability.
However, during the signal conversion process, the presence of digital resolution and linearity errors in a DAC can produce unexpected harmonic components, which may compromise signal quality. Moreover, the presence of significant harmonic components in a DAC’s output necessitates performing intricate and challenging signal processing tasks, such as filtering. Therefore, by mitigating these harmonic components, improvements can be made to a DAC’s SFDR, leading to reduced complexity and costs pertaining to signal processing.
As mentioned in [1], the SFDR of a DAC is an inherent characteristic that remains unchanged after chip fabrication due to the constraints imposed by chip design and manufacturing processes. In practical applications, the SFDR of a DAC is primarily influenced by harmonics and exhibits a high sensitivity to temperature and aging. In this context, if it were feasible to mitigate the harmonic distortion in DAC output, it could potentially enhance the SFDR to some extent, which would significantly improve the performance of communication systems.
Therefore, Chen and Gielen [2] proposed a method for enhancing DAC’s SFDR by improving the delay difference in its output. Meanwhile, Luschas and Lee [3] chose to mitigate the impact of the output mismatch on the SFDR by enhancing the DAC’s output impedance matching. Notably, although the aforementioned approach was able to improve the SFDR to a certain extent, it lacked flexibility because it was typically implemented during the early stages of DAC design. Researchers have also applied the predistortion technique to address nonlinearity in DAC [4]. However, automatic suppression of the harmonics generated by different frequencies and environments could not be achieved. Consequently, prolonged development cycles and increased costs have hindered the quest for SFDR improvement.
In this article, we propose a cost-effective and versatile method for harmonic suppression in DACs, as depicted in Fig. 1. Specifically, we employed software algorithms to suppress harmonics. First, the PSO-PID (particle swarm optimization proportional-integral-derivative) algorithm was used to identify an optimal signal for harmonic suppression. Following this, digital predistortion technology was employed to compensate for any resulting harmonic distortion within the DAC, thereby achieving the objective of attaining an enhanced SFDR.
This article is organized as follows: Section II offers a comprehensive explanation of the underlying principles governing the adaptive algorithm PSO-PID and the Hilbert transform-based memory polynomial (HTBMP) model while also outlining the methodology for parameter extraction when using the HTBMP model. This is followed by Section III, which details the experimental verification conducted in this study and analyzes the obtained results. Finally, the concluding chapter summarizes the findings of this paper.
II. Principle of Adaptive Algorithms for Harmonic Suppression
1. Principle of the PSO-PID Algorithm
The PID algorithm is a feedback control algorithm that has been widely employed in the field of industrial control owing to its simple structure, excellent stability, reliable operation, and convenient adjustability. The fundamental concept that defines this algorithm involves adjusting the system’s control input based on the disparity between the current desired output and the actual output of the system by conducting proportional, integral, and derivative control actions. This adjustment aims to minimize deviations between the system output and the desired value as much as possible by appropriately setting the proportional, integral, and derivative coefficients. Consequently, it ensures a rapid response time, offers accurate tracking capabilities, and improves overall system stability.
Furthermore, in contrast to the proportion control-based negative feedback iterative method, the PID algorithm is capable of eliminating steady-state errors [6]. Steady-state error refers to the discrepancy between the desired output value and the actual output value when the system control process approaches stability. Notably, the integral component plays a primary role in reducing steady-state error and enhancing system accuracy. The effectiveness of an integral action depends on the integration time constant—a larger integration time constant results in weaker integral action, and vice versa. Meanwhile, the derivative component reflects deviation signal trends (i.e., the rate of change). Furthermore, it introduces an early correction signal into the system before deviation signals become too large, thereby accelerating system response times and minimizing adjustment periods.
Considering this context, the relationship between the expected output from a system which denoted as ỹ(n), and the actual output is y(n), referred to as e(n), can be expressed as Eq. (1):
Therefore, according to the definition of the PID algorithm, it can be expressed as Eq. (2):
where kp, ki, and kd are the coefficients for the proportional, integral, and derivative terms, respectively.
As previously mentioned, the appropriate selection of PID parameters plays a crucial role in a negative feedback system. The estimation methods for these parameters can be categorized into traditional approaches and intelligent algorithms. Common traditional methods include empirical techniques, trial-and-error procedures, and root locus analysis. These algorithms rely heavily on manual expertise, involve time-consuming trial-and-error iterations, lack global search capabilities, necessitate high accuracy of the system model, and exhibit a limited applicability range.
Intelligent algorithms, on the other hand, are capable of conducting global searches, which enables them to avoid being trapped in local optima and find a global optimal solution or a solution close to it for the PID parameters. Furthermore, they can optimize parameters automatically, thus reducing manual intervention and improving the efficiency and accuracy of parameter optimization. Additionally, they exhibit strong adaptability and a considerable degree of generalization ability, which allows them to quickly identify suitable PID parameters for different control systems, thereby enhancing control performance. Moreover, intelligent algorithms can be easily implemented in parallel computing, where they take full advantage of the multicore processors of modern computers to improve computational speed.
As shown in Fig. 2, in this study, we employed the PSO algorithm to optimize the PID parameters. PSO is a swarm intelligence-based optimization algorithm that emulates the foraging behavior exhibited by birds to identify the optimal solution for a given problem. As a result, the PSO algorithm effectively handles continuous nonlinear problems [7]. Originally proposed by Eberhart and Kennedy [8], this approach conceptualizes the entire population as a swarm, where each particle explores a predefined search space to attain its individual best solution while also collectively interacting with other particles to achieve global solutions. This algorithm comprises displacement and velocity equations, which are influenced by factors such as weight, personal best coefficient, global best coefficient, number of particles, and number of iterations.
In this heuristic search technique, each agent examines its position information within the solution space to update the personal best displacement of its optimal solution, which in turn updates the global best displacement to achieve the globally optimal solution. Meanwhile, the velocity components are calculated based on the displacement of each particle. The coefficients or acceleration constants c1 and c2, representing the velocity of the particles toward their individual and global best positions, respectively, determine the speed and displacement of the particles. In this context, it is crucial to maintain a balance between the constants, since a higher value would compel the particles to rapidly converge toward or surpass the target position, while a lower value may result in overshooting of the target coordinates during exploration without proper recall. Notably, the inertia value ω governs the exploration rate between an individual’s best position and the global position.
Eqs. (3) and (4) present the fundamental velocity and displacement formulas for standard swarm particles:
where ω is inertia weight; c1, c2 are non-negative constant acceleration;
To improve the balance between the global search and local search capabilities of the algorithm, Shi and Eberhart [8] proposed using a linearly decreasing inertia weight, as depicted in Eq. (5):
In this equation, ωstart represents the initial inertia weight, ωend denotes the inertia weight at maximum iteration count, and Tmax signifies the maximum number of iterations. Usually, optimal performance is achieved by the algorithm on setting ωstart = 0.9 and ωend = 0.4.
The flowchart in Fig. 3 illustrates the procedural steps involved in PSO. Initially, the positions and velocities of the particles within a population are randomly initialized, following which the fitness value of each particle is computed to find the individual and global optima. Based on these optima, the positions and velocities of the particles are updated. This iterative process continues until a termination condition is met. Notably, each particle represents three parameters of a PID controller: kp, ki, and kd. Furthermore, the fitness evaluation of each particle involves calculating the normalized mean square error (NMSE) between the actual output and the distortion-free output obtained through the PID negative feedback iteration. The NMSE is then compared against a predefined threshold to determine whether it satisfies the specified condition. If it fails to meet this condition, the particle swarm is updated. The program terminates either when the NMSE meets the set condition or when it reaches the maximum number of iterations.
2. Accuracy Evaluation Function
The NMSE is a robust approach for assessing data fitting. It quantifies the accuracy of fitting outcomes based on root mean square deviations. As a result, the NMSE is considered the accuracy evaluation function in this paper. Eq. (6) presents the computational formula for NMSE:
Here, k denotes the k -th iteration and N represents the length of the signal. A smaller NMSE value indicates a more accurate fitting effect of the data. Furthermore, yk(n) refers to the actual output of the model, whereas ỹk(n) represents the fitted value of its output.
3. Accuracy Evaluation Function
The introduction of harmonic distortion into the output signal by DAC during the process of converting digital signals to analog ones can be attributed to the non-ideal characteristics of devices, such as nonlinearity, limited bandwidth, and noise. To mitigate the generation of harmonics, a digital predistorter can be cascaded prior to feeding the signal into the DAC for preprocessing. In this regard, traditional memory polynomials [9] can be expressed as Eq. (7):
where q represents the memory term, k denotes the nonlinear order, and ckq signifies the model coefficients.
Notably, in Eq. (7), the k-th order higher-order term xk(n) of the input signal encompasses not only the k-th harmonic components and their associated sum and difference frequency components but also the lower-order harmonics located in proximity and their corresponding sum and difference frequency components. As illustrated in Fig. 4, when the input signal x(n) is a two-tone signal, x6(n) generates the fourth harmonic, second harmonic, and related spectral components. Similarly, x4(n) also produces second harmonic components. However, if we intend to modify only the second harmonic component, it becomes necessary to adjust x2(n), x4(n), and x6(n). Furthermore, altering x4(n) and x6(n) would inevitably impact both the fourth harmonic and the sixth harmonic, which would compel us to deviate from our intended objective. Hence, one of the primary focuses of this paper is to modify x6(n) while minimizing any influence on lower-order harmonics.
First, we applied the Hilbert transform [10] to the real-valued signal x(n), as shown in Eqs. (8) and (9):
where * denotes convolution and j represents the imaginary unit. We transformed x(n) into a complex-valued signal z(n), containing pure k-th harmonics along with the surrounding parasitic components. Fig. 5 illustrates the spectra of z(n), z2(n), z3(n) and z4(n) for a two-tone signal. It is observed that modifying z4(n) does not affect the fourth or second harmonic compared to x4(n). However, the lower-order components are missing in z4(n), which requires compensation.
We exploited the spectral displacement of zk(n) to align it with the k-th power term of the lower-order components of x(n). In digital communication systems, the center frequency fc and bandwidth information of a digital signal sent to a DAC are often known a priori. Therefore, by applying complex conjugation to the center frequency, we obtained −fc in the negative frequency domain, resulting in a digitally sampled −fc(n). Following this, we calculated the spectrum shift using Eq. (10):
where m represents the number of shifts.
By multiplying zk(n) with fc,m(n), Eq. (11) demonstrates the completion of the frequency spectrum shift for zk(n). Following this shift, we obtained the low-order harmonic components of xk(n), along with the sum and difference frequency components surrounding these harmonics. Notably, to differentiate it from zk(n), we denote the result obtained after the frequency spectrum shift as zk_l(n). Fig. 6 presents the simulation results obtained using z6(n) of a two-tone signal as an example.
Using Eq. (9), we calculated zk(n) and then multiplied it with Eq. (10) to obtain zk_l(n), as depicted in Eq. (11). Subsequently, by optimizing xk(n) in Eq. (7), we formulated Eq. (12). In this context, it is important to note that the optimization process for the memory term follows a similar approach as that for the current term. However, elaborating on this further is beyond the scope of this study. Notably, Eq. (12) reveals that xk(n – q) exhibits a higher degree of parameterization:
By substituting Eq. (12) into Eq. (7) and applying the Hilbert transform to y(n), we obtained Eq. (13), which represents the HTBMP model. The flowchart for this model is illustrated in Fig. 7.
The original signal was used as the input for the predistorter, while the PSO-PID algorithm was employed to obtain the optimal input signal xd(n) for achieving a harmonic-free DAC output. Once sufficient accuracy was achieved, the optimized signal was used as the output of the predistorter. This can be expressed as Eq. (14), where a⃗ represents the parameter vector of the HTBMP.
Where
Subsequently, the parameter vector a⃗ of the HTBMP model was extracted using the least squares method proposed in [11], formulated as Eq. (16):
III. Experimental Validation for Harmonic DPD using the PSO-PID Adaptive Algorithms
1. Testing Setup and Verification of the PSO-PID Harmonic DPD Algorithm
We chose to conduct experimental verification in the shortwave frequency band due to its broad frequency range, spanning 1.6 MHz to 30 MHz, and the current high sampling rate of DACs that enables the efficient identification of harmonics within this range. Enhancing the SFDR of DACs in shortwave communication systems can effectively mitigate system nonlinear distortion and improve system stability, ultimately enhancing the reliability of shortwave communication. Furthermore, given that shortwave communication necessitates coverage across multiple frequency bands, increasing the SFDR of DACs will not only alleviate the pressure on analog filter groups but will also ensure higher clarity and integrity during signal transmission, in turn extending the reach of shortwave communication. Improving the SDFR of DACs may enhance interference resistance in complex electromagnetic environments while maintaining stable communication. Therefore, advancements in DAC’s SFDR have the potential to optimize spectrum resource utilization and improve the communication efficiency of shortwave communication systems.
In this study, we employed a methodology involving the measurement and averaging of multiple sine wave signals within an extended dynamic range to assess the SFDR of DACs. Experimental verification was conducted using single-tone sine wave signals. By capturing real-time data of the DAC input and output, we simulated the PSO-PID algorithm, as depicted in Fig. 8. Subsequently, a computer-generated predistorted signal based on the PSO-PID algorithm was generated and fed into an arbitrary function generator. The digital signal from this generator was then converted into an analog signal, which underwent further digitization by a digital oscilloscope for spectrum analysis. The resulting data were iteratively processed using the PSO-PID algorithm. In this context, it should be noted that during the data acquisition process, the signal coupled from the DAC output had to be adjusted to a suitable magnitude before being fed into the analog-to-digital converter (ADC) so as to let the ADC work in an ultra-linear region. Hence, the distortions caused by ADC can be ignored. The distortions included in the captured data were mainly caused by the DAC. In addition, we used a spectrum analyzer with a dynamic range of 116 dB to verify the effect of the PSO-PID HTBMP.
The actual fabricated DAC data acquisition platform is presented in Fig. 9. Initially, a sine signal was generated using MATLAB and then imported into Keysight M8190A arbitrary waveform generator. The clock of the arbitrary waveform generator was configured to internal mode, with the output level set to 600 mV. For collecting the DAC output data, Keysight M9703B digitizer and Keysight 89600VSA software were employed. To ensure synchronization between the arbitrary waveform generator and the digitizer in terms of the clock source, the digitizer’s clock was set to synchronize with the output clock of the arbitrary waveform generator. Similarly, to synchronize their sampling times, the arbitrary waveform generator was set as the trigger source for the digitizer.
After acquiring the DAC input and output data, a negative feedback iteration using PSO-PID was performed. The PID parameters were constrained within the range of −1 to 1, with the particle swarm comprising 30 particles. Both the acceleration factors, c1 and c2, were set to 0.8, while the maximum number of iterations for the particle swarm was limited to 60 times. Fig. 10 illustrates the simulation results of the PSO-PID parameters and NMSE variation with respect to the number of iterations. The graph shows that the PID parameters tended to stabilize and exhibit excellent convergence performance in terms of NMSE after 40 iterations. By employing the PSO-PID algorithm, we obtained three optimal parameters for the PID, as well as the harmonic distortion compensator’s output signal xd(n). Finally, by utilizing both the original signal x(n) and signal xd(n) as the input and output, respectively, we calculated the parameters of the predistorter model in the HTBMP model.
2. Analysis of Test Results
After extracting the harmonic digital predistortion (DPD) parameters for the DAC, the original signal was fed into the DAC equipped with a predistorter, and the resulting output signal was collected for harmonic spectrum analysis. Fig. 11 illustrates the suppression effect on harmonics in the case of a typical single-tone sinusoidal signal, demonstrating that the digital predistorter effectively mitigated the harmonic components originating from the DAC.
The effects of DPD on single-tone sine wave signals at frequencies of 2 MHz, 6 MHz, 12 MHz, 18 MHz, and 24 MHz are recorded in Table 1. The findings presented in Table 1 point to a minimum improvement of 9 dB in the second harmonic, indicating an increase in the DAC’s SFDR by at least 9 dB. Notably, improvement in the third harmonic was not obvious and was far less than in the second harmonic.
Fig. 12 traces the improvements in the DAC’s harmonics corresponding to different frequencies, as well as the improvements in its SFDR. With an increase in frequency, both harmonics and SFDR fluctuate to a certain extent. It is evident that the PSO-PID HTBMP DPD effectively suppressed the DAC’s harmonics and improved its SFDR at different frequencies.
IV. Conclusion
The present study proposes a PSO-PID algorithm-based HTBMP predistortion model for harmonic suppression in DAC. The principles of the PSO-PID algorithm and the HTBMP model were theoretically analyzed. The simulation and experimental verification results highlight that the HTBMP harmonic DPD base on the PSO-PID algorithm effectively enhanced the SFDR of a DAC by at least 9 dB.
Acknowledgments
This work was partially supported by the National Natural Science Foundation of China (No. 62371266, 62071264, and U1809203) and the Key Research Project funded by China Tobacco Zhejiang Industrial Co. (No. ZJZY2023E001).
References
Biography
Junshi Lv, https://orcid.org/0000-0002-1287-6881 received his B.S. degree in electrical and automation engineering from Sanjiang University, Nanjing, China, in 2014. In 2020, he received his M.S. degree from the Faculty of Electrical Engineering and Computer Science, Ningbo University, Ningbo, China, where he is currently pursuing his Ph.D. His main research interests pertain to the field of wireless communication, especially high-efficiency radio frequency (RF) power amplifier (PA) design, analog and digital predistortion (DPD), and nonlinear modeling.
Chengting Zhang, https://orcid.org/0009-0001-8661-6281 graduated from the South China University of Technology with a major in communication engineering in 2008. He went on to attain his master’s degree in network engineering from Zhejiang University in 2010. At present, he works as a senior engineer for network administration at China Tobacco Zhejiang Industrial Co., Ltd., with his main research focus being computer communication technology and network security.
Honglv Wang, https://orcid.org/0009-0008-1576-9409 is a senior engineer currently working with China Tobacco Zhejiang Industrial Co. Ltd. He is mainly engaged in tobacco informatization research and enterprise digital transformation.
Gaoming Xu, https://orcid.org/0000-0001-5466-8215 received his B.S. degree in electronic and information engineering from the Naval Aeronautical Engineering Academy, Qingdao, China, in 2007, and his M.S. and Ph.D. degrees from the Faculty of Electrical Engineering and Computer Science, Ningbo University (NBU), Ningbo, China, in 2010 and 2015. He is currently working as an associate professor in the Faculty of Electrical Engineering and Computer Science at NBU. His main research interests are in the area of wireless communication, with a focus on high-efficiency RF power amplifier design, digital predistortion, analog predistortion, and nonlinear modeling.
Taijun Liu, https://orcid.org/0000-0002-2582-5290 received his B.S. degree in applied physics from the China University of Petroleum, Dongying, China, in 1986; his M.Eng. degree in electrical engineering from the University of Electronic Science and Technology of China, Chengdu, China, in 1989; and his Ph.D. degree from École Poly Technique de Montréal, Université de Montréal, Montreal, QC, Canada, in 2006. He is currently working as a professor in the Faculty of Electrical Engineering and Computer Science at Ningbo University. His current research interests include nonlinear modeling and linearization for wideband transmitters/power amplifiers, intelligent IoT, intelligent wireless sensing, and ultra-linear high-efficiency intelligent power amplifiers for broadband wireless and satellite communications systems.