ABSTRACT
A sub-nanosecond clock synchronization scheme based on the field programmable gate array (FPGA) is proposed for the Fiber Channel (FC) communication system in this paper. The counter value of the slave node is synchronized to that of the master node through the embedded IEEE 1588 protocol over the communication link. In order to ensure the counter clocks have the same frequency in both nodes, which is recovered from the FC communication link, the clock phase difference is measured by the digital dual mixer time difference technique and the data recovery technique in the Gigabyte Transceiver, and then it is compensated by the mixed-mode clock manager in FPGAs. The proposed clock synchronization approach is evaluated with an FC communication system that has a serial rate of 12.5 Gbps, and the reported experimental results show that the proposed clock synchronization module can achieve a time difference lower than 1 ns.
ABSTRACT
The stability and accuracy of the clock signal are crucial for the proper operation of various electronic devices and systems, as they directly impact system performance. In high-speed electronic systems, the clock signal is susceptible to interference by crosstalk. Therefore, evaluating the performance of the clock signal under crosstalk disturbance is important. Jitter, commonly used as an indicator to assess the degree of this interference, plays a significant role in this evaluation. In this paper, a method is proposed for assessing crosstalk-induced jitter (CIJ) based on scattering parameters. To verify the effectiveness of the method, CIJ was measured for clock signals with frequencies of 25, 100, and 156.25 MHz. In addition, the experimental results are well in agreement with the theoretical model. Thus, the potential application of this method is to assess the performance of circuits or electronic systems.
ABSTRACT
Direct-conversion receivers (DCRs) have been widely used in recent years due to their small size and low power consumption. However, the mismatch between the in-phase (I) and the quadrature (Q) branches will seriously affect the performance of the DCRs. This paper proposes a novel blind compensation method to suppress the interference introduced by IQ mismatch. Based on the Hilbert transform, our proposed method can obtain the orthogonal signal of the I-channel signal by utilizing the Weaver architecture. Compared with traditional compensation methods, the main difference of the proposed method is that it ignores prior information, training sequences, and additional hardware circuits. Furthermore, the complexity of the proposed blind compensation method is low because no iterative operations are involved in the compensation process. The simulation results show that the proposed method has an excellent compensation performance, especially in wideband applications.
ABSTRACT
Frequency measurement is one of the key techniques in high-precision data acquisition technology of broadband signals. Generally, frequency measurement not only needs to deal with a large amount of data processing but also requires a high precision, but these two aspects are sometimes difficult to reconcile. Some algorithms are overly dependent on the accuracy of the to-be-measured data, which might not be the desired option for real projects since it is almost impossible to get ideal error-free data. This article adopts a frequency measurement method based on the coordinate rotation digital computer algorithm, differential algorithm, and Kalman filter. The use of these algorithms for the frequency measurement process would not only simplify the calculation but also reduce the effect of the measurement error. This method can measure all signals that satisfy the sampling theorem and can also measure multi-channel parallel signals. The experimental results of data simulation and actual measurement on the hardware platform show that the accurate frequency measurement algorithm has a strong data processing ability, stable measurement, and steady improvement in the accuracy of measurement results, which can meet the needs of most instruments for accurate frequency measurement. The measurement error could be reduced to the percentile by the Kalman filter and could be reduced to below the thousandth by the combining the algorithms.
ABSTRACT
In this paper, a novel algorithm called two-dimensional sliding fast Fourier transform (2D SFFT) algorithm is proposed. This algorithm organizes one-dimensional data in two dimensions and calculates the spectrum of current data by using the existing spectrum and new collected data. The algorithm formula and accurate simulation results show the following: first, the computation required by the proposed 2D SFFT algorithm is lower than that required by the traditional sliding discrete Fourier transform algorithm when the sliding rate is larger than or equal to 4/M, where M is the sequence length. Moreover, the computation required by the proposed 2D SFFT algorithm is lower than that required by the fast Fourier transform (FFT) algorithm when the sliding rate is less than or equal to 6.25%. Finally, the error between the spectrum calculated by the 2D SFFT and FFT algorithms is less than 10-10. The 2D SFFT algorithm is used to increase the power of the ultra-short pulse, which is initially invisible in the frequency-domain window of the mixed-domain oscilloscope. Therefore, the 100% probability of intercept of the mixed-domain oscilloscope is lower.
ABSTRACT
For wideband receiver systems, it is challenging to compensate the in-phase/quadrature (I/Q) phase mismatch by traditional methods, especially with a time delay deviation (TDD) between the I/Q channels. Considering the above situation, this paper proposes a full-scale I/Q phase imbalance model concerning TDD. The model divides phase mismatch into two parts, i.e., the linear phase (LP) part and the nonlinear phase part, and compensates each part with the corresponding compensation module separately. The design strategy of the compensation module is innovatively transformed into a constrained nonlinear optimization problem, and a metaheuristic algorithm, the flower pollination algorithm (FPA), is utilized to be the optimizer. The results of the contrast simulation with the LP elimination method show the efficiency of the proposed method. In addition, the superiority of the FPA-based structure is verified by comparing with other metaheuristic algorithms, the artificial bee colony technique, the bat algorithm, and the differential evolution algorithm, in terms of the compensation accuracy, algorithm stability, runtime consumption, and convergence performance. Ultimately, the image rejection ratio improvement on the actual platform after compensation is measured, which validates the proposed compensation structure and the corresponding optimization method practically, and the FPA is still the best choice among the competent optimizers.
ABSTRACT
The implementation of high-performance and high-frequency temperature-compensated crystal oscillator (HFTCXO) still faces great challenges. In this article, a new temperature-compensation method for HFTCXO with closed-loop architecture is described. The sample of 100-MHz HFTCXO with the measured temperature stability of ±0.22ppm/-40 °C ~ +85 °C and the phase noise of -151 dBc/Hz@1 kHz and -163 dBc/Hz@10 kHz was designed. Experimental results show that this method can realize real-time high-precision temperature compensation of high-frequency crystal oscillator.
ABSTRACT
Electromagnetic interference (EMI) measurement is essential in wireless communications to avoid both performance degradation and serious fault failures. In this paper, we present the design of an acquisition system for a time-domain EMI measurement receiver intended for monitoring potential EMI. The platform consists of both hardware and software systems. The hardware system is constructed using a modular design method and contains a conditioning module, a data acquisition and processing module, and a clock generation module. The software system provides an interactive interface for device operation and offers an automatic sampling/processing procedure that is developed on a slot 0 controller. An additional outstanding feature of the proposed system is its high dynamic range. We propose a novel multi-stage architecture and a data reconstruction algorithm that are combined to achieve large dynamic range acquisition. The proposed acquisition architecture breaks through the dynamic range limitations of conventional analog-to-digital converters (ADCs) on a significant scale. The implemented system is able to capture an analog waveform with a 300 MHz bandwidth and a 1 GS/s sampling rate. The experimental results show that the maximum dynamic range achieved is 96.43 dB, which is far greater than the single ADC dynamic range of 52 dB.
ABSTRACT
DC offsets in a quadrature mixing structure will affect the system performance. This paper proposes a Hilbert transform-based algorithm to blindly calibrate the DC offsets by filtering the inphase and quadrature signals with the same Hilbert filters. Compared with the traditional averaging-based method, this method consumes less time and resources. Moreover, the proposed method need not interrupt analog-to-digital converter's conversion, featuring the characteristics of high efficiency and good real-time performance. The Matlab simulation results verify the correctness of the proposed algorithm, and finally, an experimental platform is designed in detail to verify the practicability of this blind calibration algorithm.
ABSTRACT
The main factors that enable capture of complex and transient signals in real-time are improved sampling rates and processing speeds. The time-interleaved architecture is an effective method that allows systems to break through the speed bottleneck of single analog-to-digital converters (ADCs) and go beyond the state-of-the-art process technology limit. However, the performance of the acquisition system may be reduced because of the offset, gain, and time mismatch errors that occur in time-interleaved ADC systems. To correct these errors, this paper first proposes a self-adaptive correction algorithm and then introduces real-time solutions for this algorithm. Finally, the proposed calibration method is implemented in a digital phosphor oscilloscope. Simulations and experimental testing indicate that this system shows good real-time performance and provides a high dynamic performance with an effective number of bits of 7.3 bits and a signal-to-noise ratio of 45.5574 dB.
ABSTRACT
The wideband signal transceiver module forms the core component of the radar signal target simulator and the key module in measuring the Doppler shift. In this paper, we present a design for a miniaturized wideband signal transceiver module based on a PXIe bus, with an operating frequency range of 65 MHz-3 GHz, an instantaneous measurement bandwidth of 800 MHz on a standard 3U (where 1U = 4.445 cm) board with three slots. The module adopts a zero intermediate frequency transceiver, and the baseband uses two analog to digital converters, a field programmable gate array, and two digital to analog converters as the core hardware architecture. The module combines key technologies including variable-frequency local oscillator synthesis, radio-frequency broadband signal conditioning, and high-speed high-resolution baseband signal acquisition and generation. This paper describes a novel design for the Doppler shift function. Considering the limited size and volume of the module, Doppler shift technology combining a complex signal and a real signal was adopted to achieve full band coverage with an operating frequency range of 65 MHz-3 GHz. We verified that this module could generate the Doppler shift in the frequency band from -400 Hz to 400 Hz in steps of 2 Hz and with an error of ±2 Hz.
ABSTRACT
Memory depth represents an oscilloscope's capability to continuously acquire and store sampling points at the highest real-time sampling rate. Improving the memory depth is conducive to improving the system's capability to analyze waveform details, but it will slow down the response time of the system. This paper proposes a high-speed deep memory data acquisition system with a real-time sampling rate of up to 10 GSPS and a memory depth up to 1 Gpts based on ultra-high-speed parallel sampling, waveform quick positioning and zooming, and waveform 3D mapping and display. This study focuses on the use of waveform quick positioning and zooming techniques under deep memory conditions to solve the slow response time and low waveform capture rate that are the result of improving the memory depth. This paper gives the experimental results for real-time signal data acquisition. The results show that adopting waveform quick positioning and zooming techniques achieved the following aims: significantly improved system response time, rapid positioning and display waveform details available to users, fast waveform capture rate and improved oscilloscope performance.
ABSTRACT
The acquisition of waveforms and the analysis of transient characteristics of signals are the fundamental tasks for time-domain measurement, while the reduction of the measuring gap till seamless measurement is extremely important to the acquisition, measurement, and analysis of transient signals. This paper, aimed at the seamless time-domain measurement of non-stationary transient signals, proposes an approximate entropy-based characteristic signal extraction algorithm on the basis of information entropy theories. The algorithm quantitatively describes the complexity (amount of information) of sampled signals using the approximate entropy value, self-adaptively captures characteristic signals under the control of the approximate entropy in real time, extracts the critical or useful information, and removes redundant or useless information so as to reduce the time consumption of processing data and displaying waveforms and realize the seamless time-domain measurement of transient signals finally. Experimental results show that the study could provide a new method for the design of electronic measuring instrument with seamless measurement capability.
ABSTRACT
The digital channelization technology has been applied in many electronic areas, and the real-time broadband spectrum analysis has been the research hotspot in the area of signal processing. This paper introduces the channelized broadband signal spectrum analysis method. Based on the weighted overlap-add (WOLA) structure, this method divides the input broadband signal into several sub-bands or channels, and then downconverts and decimates the sub-band signals to obtain the baseband signals with a low sampling rate. The spectrum analysis results of the input broadband signal are achieved by conducting further decimation, fast Fourier transform and spectrum splicing to the baseband signals. The Matlab simulation results verify the correctness of the WOLA structure, and finally, an experimental platform is designed in detail to verify the practicability of this broadband spectrum analysis method.
ABSTRACT
This paper proposes an algorithm to estimate the channel mismatches in time-interleaved analog-to-digital converter (TIADC) based on fractional delay (FD) and sine curve fitting. Choose one channel as the reference channel and apply FD to the output samples of reference channel to obtain the ideal samples of non-reference channels with no mismatches. Based on least square method, the sine curves are adopted to fit the ideal and the actual samples of non-reference channels, and then the mismatch parameters can be estimated by comparing the ideal sine curves and the actual ones. The principle of this algorithm is simple and easily understood. Moreover, its implementation needs no extra circuits, lowering the hardware cost. Simulation results show that the estimation accuracy of this algorithm can be controlled within 2%. Finally, the practicability of this algorithm is verified by the measurement results of channel mismatch errors of a two-channel TIADC prototype.
ABSTRACT
In traditional digital storage oscilloscope (DSO), sampled data need to be processed after each acquisition. During data processing, the acquisition is stopped and oscilloscope is blind to the input signal. Thus, this duration is called dead time. With the rapid development of modern electronic systems, the effect of infrequent events becomes significant. To capture these occasional events in shorter time, dead time in traditional DSO that causes the loss of measured signal needs to be reduced or even eliminated. In this paper, a seamless acquisition oscilloscope without dead time is proposed. In this oscilloscope, three-dimensional waveform mapping (TWM) technique, which converts sampled data to displayed waveform, is proposed. With this technique, not only the process speed is improved, but also the probability information of waveform is displayed with different brightness. Thus, a three-dimensional waveform is shown to the user. To reduce processing time further, parallel TWM which processes several sampled points simultaneously, and dual-port random access memory based pipelining technique which can process one sampling point in one clock period are proposed. Furthermore, two DDR3 (Double-Data-Rate Three Synchronous Dynamic Random Access Memory) are used for storing sampled data alternately, thus the acquisition can continue during data processing. Therefore, the dead time of DSO is eliminated. In addition, a double-pulse test method is adopted to test the waveform capturing rate (WCR) of the oscilloscope and a combined pulse test method is employed to evaluate the oscilloscope's capture ability comprehensively. The experiment results show that the WCR of the designed oscilloscope is 6,250,000 wfms/s (waveforms per second), the highest value in all existing oscilloscopes. The testing results also prove that there is no dead time in our oscilloscope, thus realizing the seamless acquisition.
ABSTRACT
This paper proposes an instantaneous burst carrier frequency measurement scheme combined with timestamping counting method based on field programmable gated array. With the multiplication and phase shift of counting clock, multiple parallel counters run continuously, and the times of rising edge of measured frequency f(x) are counted simultaneously. The frequency of f(x) is calculated using the least squares line fitting method of regression analysis from the count values of f(x) and timestamping clock. Experiment results show that the proposed approach can effectively reduce the error introduced by the quantization error of ±1, improve measurement accuracy by about 2 to 3 digits, reduce measurement uncertainty by more than 20%, and diminish the processing time to 200 ns compared to traditional methods under the same measurement conditions.