1. Introduction
The analog-to-digital converter is a bridge that is applied to convert analog signals into discrete signals [
1]. It is widely used in various fields, such as wireless communication, electronic radar, medical equipment, etc. With the progress of the time, the demand for ADC is growing. The essential criteria for ADC are higher precision, higher speed, lower power consumption, etc. It is difficult for a single-structure ADC to incorporate all these three factors in the existing technologies. Fortunately, TIADC can solve this problem effectively [
2]. The TIADC can double the sampling rate of the whole ADC, and it can easily reach the sampling rate of 1 GS/s [
3].
The TIADC is composed of M sub-channels. Due to the manufacturing process, layout routing, temperature, voltage, and other such factors, mismatches between the sub-channels will occur, such as offset mismatch, gain mismatch, and timing mismatch [
3]. Indeed, the degradation of the Signal-to-Noise Ratio (SNR) and Spurious Free Dynamic Range (SFDR) of the output signal are most probably connected to these mismatches. The system as a whole cannot satisfy the demands for high precision and high speed unless these errors are calibrated accurately. The common calibrations for offset and gain mismatches involve the accumulation and averaging of the input signal, and this is followed by its subtraction from the original signal. This approach is both straightforward and efficient [
4]. The calibration of timing mismatch is the most challenging compared to other mismatches [
2]. Therefore, the architecture proposed in this paper was mainly designed to solve the timing mismatch problem.
The calibration of timing mismatches can be accomplished through methodologies that are akin to those employed in analog loop calibration. The analog loop calibration method primarily involves filtering the TIADC sampling clock, and it is coupled with the adjustment of clock phases through the observation of output discrepancies in the digital domain for each sub-ADC channel. This procedure serves to effectively eliminate sampling timing mismatches [
5,
6]. Ref. [
5] introduced a kind of foreground calibration technology based on an analog auxiliary circuit, which predominantly depended on the filtration efficacy of the analog domain filter circuit with respect to the clock signal. However, this method exhibited significant limitations, and its calibration performance was greatly influenced by temperature variations, process variations, and voltage fluctuations. Ref. [
6] introduced a coarse adjustment technique for an analog variable delay line. Nonetheless, the fixed and coarse step sizes in this method imposed limitations on its effective adjustment range, as discerned from experimental analysis.
In contrast, all-digital calibration technology offers high reliability and powerful portability advantages, therefore effectively addressing the aforementioned issues. Several methodologies have been developed to address timing mismatch calibration within an all-digital domain. The investigations of all-digital calibration structures are mainly focused on two parts: the estimation method and the compensation method.
Thereinto, the predominant error estimation methodologies entail the utilization of the cross-correlations among sub-channels [
2,
3,
7,
8,
9,
10,
11,
12], or the introduction of a reference channel [
13,
14,
15,
16]. Based on the correlations among sub-channels, a complex computational matrix is commonly constructed. For example, Ref. [
9] introduced an estimation methodology that employs signal modulation for the construction of an error coefficient matrix. Nevertheless, a notable limitation was observed in terms of the sluggish convergence rates (110 k) and constraints associated with finite input bandwidths. Please note that these methods that are predicated on signal modulation or Hadamard matrix operations are infrequently employed in practical applications due to their elevated computational intricacy. Calibration structures that are predicated on reference channels commonly exhibit relative simplicity. The estimation process can be construed as the assessment between a designated sub-channel and the reference channel. The calibration procedure can be delineated as a specialized form of a dual-channel calibration structure, i.e., one that is impervious to the influence of channel count expansion. Notably, this configuration is frequently characterized by diminished hardware consumption. In the paradigm of sub-channel cross-correlation, the escalation in the number of channels engenders an exponential amplification in the complexity of the formulated system of error functions. This not only augments the intricacy of design but also imparts a substantial inefficiency in terms of hardware utilization. Ref. [
14] proposed an estimation methodology, one which entailed the application of a high-pass filter for the processing of input slopes that pertained to both the sub-ADC and the reference ADC. However, the reference channel necessitated a significant degree of precision, thus resulting in substantial hardware consumption with many multiplication–addition units in the usage of a high-precision filter case. Additionally, the calibration performance gradually deteriorates with increasing frequency, which is a common issue faced by most calibration structures [
3,
13,
14,
15]. Typically, increasing the order of the filter and parallel correlation derivative [
6] was employed to mitigate this deterioration, but it inevitably led to an increase in hardware consumption. Ref. [
16] proposed an innovative split-based structure. This design segregated an M-channel TIADC into two segments that featured mutually co-prime operating frequencies. At each sampling instance, each segment actively provides a sub-ADC channel as a reference for the other. The utilization of the difference in the outputs from the dual paths serves as a new thought for actively extracting the temporal misalignment information between channels. Ref. [
17] proposed a methodology for error computation based on the detection of the monotonic trends in the sampled data from sub-channels and reference channels, whereby they then subsequently applied cumulative difference accumulation. It is noteworthy that this structural approach eliminates the necessity for any multiplication or division operations. However, there have only been a few reports about timing mismatch calibration techniques based on reference channels, and their effectiveness needs further verification and improvement.
Most compensation methods rely on either filtering techniques [
18,
19,
20,
21] or Taylor expansion [
11,
22,
23]. Numerous structures of the commonly employed filter are subjects of the study, with a substantial portion of researchers directing their investigations toward fractional-delay filters and perfect-reconstruction filters. Ref. [
19] introduced a methodology that employed fractional filters for the phase adjustment in the input signal. Ref. [
20] introduced a methodology that utilized the perfect reconstruction filters to efficaciously rectify errors in reconstructed signals. However, this approach consumed high hardware and substantial power. In addition, the structures based on Taylor series expansion are highly regarded for their low complexity, rapid convergence, and notable efficacy, which make them equally popular. Ref. [
11] utilized a Taylor series expansion to approximate the error terms within the output signal. The calibration of the errors in the output signal was achieved through the consideration of the time misalignment errors and the derivatives of the input signal. The conventional approach to optimizing the output performance and the input signal bandwidth in calibration algorithms typically relies on the degree of differentiation or the second-order differential term. However, the exponential increase in the hardware consumption has not been positively linked to the commensurate improvements in the input bandwidth.
In this paper, an all-digital calibration structure is proposed to address the aforementioned issues. A reference channel is used to extract the timing mismatch of each sub-channel. The compensation structure is formulated based on the expansion of the first-order Taylor series. The derivator which is optimized with the Richardson extrapolation is based on the first-order numerical differential of the Lagrange interpolation polynomial of the 5-point formula. The optimization of the derivator circuit’s structure, as is outlined in this paper, aims to minimize the hardware consumption and the overall resource utilization. In addition, an auxiliary circuit is introduced to further enhance the output precision and performance, which improves the overall accuracy and efficiency. This design manifests attributes including the expeditious convergence, the compact hardware footprint, and the straightforward architectural framework. This paper is organized as follows.
Section 2 describes the construction of the TIADC module and the merits of the proposed calibration technique.
Section 3 shows the simulation results which verify the effectiveness of the technique.
Section 4 is mainly used to discuss the advantages and disadvantages of the proposed algorithm as well as the direction of the future work. Finally, the conclusions are drawn in
Section 5.
2. Timing Mismatch Model and Proposed Timing Mismatch Calibration Structure
Assuming that the prior calibrations with the help of a reference ADC [
24] have been made for other mismatches (i.e., offset and gain mismatches), we only need to focus on the calibration of the timing mismatch.
Figure 1 shows the structure of the M-channel TIADC with the reference channel. The output of the optimal M-channel TIADC system can be expressed as
In (
1),
is the sampled value of the input signal
at the time
, where
represents the timing mismatch of the
sub-channel and
represents the sampling period. The signal
of the entire TIADC can be expressed as
In the actual circuit, the timing mismatch is much smaller than , . represents the timing mismatch coefficient of the sub-channel.
2.1. Timing Mismatch Compensation
In practice, the timing mismatch is typically negligible in comparison to the sampling period. As a result, the first-order Taylor series is employed for approximation [
6]. The function of the timing mismatch compensation
on the
sub-channel is expressed as
where
represents the differential value of the output from the
sub-channel. The deviation coefficients of the derivator exhibit frequency-dependent variations during the execution of the differentiation operation [
25]. Therefore, the relationship between the output of the derivator (
) and the ideal derivative (
) can be mathematically expressed as
where
represents the deviation coefficient of the derivator. (
3) is rewritten as
The deviation coefficient of the derivator is solely dependent on the frequency.
Figure 2 shows the implementation architecture of the timing mismatch calibrations.
2.2. Estimation Architecture
In this sub-section, this paper presents an adaptive estimation method that utilizes the first-order statistics to address the timing mismatches. The estimation model is based on a four-channel model with an additional channel for reference. The sampling frequency of TIADC is
. The sampling frequency of the sub-channel is
. The sampling frequency of the reference channel is
. The sampling values of the reference channel are aligned with each sub-channel in turn. The sampling timing of each sub-channel ADC and the reference channel ADC is illustrated in
Figure 3.
The specific sampling sequences are extracted through down-sampling. The output sequence
of the
sub-channel after down-sampling is expressed as
The corresponding output sequence
of the reference channel needs to be extracted accordingly, which can be formulated as
where
represents the output of the reference channel, which can be expressed as
The estimation of the timing mismatch requires the utilization of the cross-correlation function between the sub-channel and the reference channel. The reference channel output sequence
(
for
,
stands for modulus function) of the corresponding sub-channel can be denoted as
the cross-correlation function
between the sub-channel and the reference channels after down-sampling, i.e.,
and
, respectively, is expressed as
where
expresses the mathematical expectation. It is realized through Modified Moving Average (MMA) [
23], which is shown in
Figure 4. It is expressed as
where
and
are the output and the input, and
represents a constant smoothing factor. The recursive calculation of the average output
relies on both the preceding result
and the current input term
. The (
11) is commonly utilized in Digital Signal Processing (DSP), and the recursive calculation of the computing moving averages. In terms of the hardware implementation, opting for a
sample period—with k being any positive integer is proved to be advantageous since multiplying the signal by
can be efficiently accomplished via the proper arithmetic shifting by k bits instead of employing a hardware multiplier.
expresses the cross-correlation of
and
. By the first-order Taylor series approximation, it could be approximately expressed as
where
is the derivative of the autocorrelation of the input signal. It could be expressed as
where
is the derivative of the reference signal. From (
10) and (
12), the cross-correlation value between
and
is expressed as
Obviously, the cross-correlation value
between
and
is expressed as
To obtain the timing mismatch, the difference of the cross-correlation values
can be derived by subtracting the two equations that is mentioned above (i.e., (
14) and (
15)):
which is in direct proportion to
.
Approximately, the timing mismatch
can be represented as
The implementation architecture of (
17) is shown in
Figure 5.
2.3. Derivator Architecture
The traditional derivator is essentially a filter, which consists of the multiplier and the delay register units. Considering that the differentiation effect of the derivator increases with its order, it inevitably consumes a large amount of the aforementioned filters to obtain a sufficiently high order (e.g., [
2,
6]) and meet the requirements of the design. This paper proposes a simplified structure of the derivator. According to the Lagrange interpolation polynomial, the first-order numerical differentiation can be expressed as [
26]
where
represents the output of the derivator. By the usage of the Taylor series expansion, the terms
,
,
, and
can be effectively expanded. Consequently, (
18) can be elegantly reformulated as
where
is the derivative of
, similarly,
and
are the fifth-order and seventh-order derivative of
, respectively. To enhance the differentiation effect of the derivator, the Richardson extrapolation is employed to construct the higher-order approximation formulas [
27]. It estimates the value of the target function by approximation with different step lengths. The first order numerical differential formula
from a higher-precision extrapolation can be expressed as
where
h represents the sampling step and
m represents the number of the iterations. As the number of the iterations increases, the output of the derivator gradually converges to the ideal derivative value. The following expression for the output of the derivator
can be derived by the single iteration:
where
is expressed as [
26]
Subsequently,
can be obtained from the Taylor series expansions of
and
, as is described below.
By comparing (
19) and (
23), the application of the Richardson extrapolation formula eliminates a certain amount of higher-order derivatives (e.g.,
) and enhances the accuracy of the derivator so that the compensation performance can be improved effectively. By substituting (
18) and (
22) into (
21), the expression can be derived as follows:
The implementation architecture of (
24) is shown in
Figure 6.
2.4. Derivator Calibration
The linearity of the derivator gradually deteriorates with the increase of the frequency. The non-negligible deviation between the actual and ideal output of the derivative significantly impacts the accuracy of the final compensation and calibration. To compensate for the output deviation at the high-frequency cases, this paper presents a straightforward structure for the calibration of the derivator.
The deviation coefficient of the derivator is solely dependent on the frequency and is unrelated to any channel. Take Channel 3, for example. The sequence
after down-sampling is selected. It is aligned with Channel 3, which is written as
Similarly, the sampling sequence
that is corresponding to Channel 3 is expressed as
The timing mismatch has been rectified after the compensation with the Taylor series, whereas the derivator still exhibits a frequency-dependent bias coefficient. The corresponding output of the derivator
is written as
Using Taylor series expansion and combining (
25) and (
27), the following results can be derived as
The expectation of the deviation coefficient can be expressed as
where the expectation of
,
and
are constants on account of the input signal
that is band-limited. The calibration architecture of the derivator’s deviation according to (
29) is shown in
Figure 7.
4. Discussion
Table 1 presents the other four works on the all-digital calibration techniques that have been proposed in recent years. Compared to the works of other groups [
9,
12], this paper presents several significant advantages over the structure with the reference channel, especially in the area consumption and the convergence speed. Compared to Ref. [
28], it achieves superior calibration results while maintaining the same level of the accuracy. A significant improvement in calibration performance for the multi-frequency signals is also observed. Compared to Ref. [
9], this paper presents a faster convergence speed and wider effective bandwidth. As mentioned, the techniques in Refs. [
9,
28,
29] fail to maintain high-precision output at the high frequencies, which result in a noticeable degradation in the calibration performance. By contrast, the proposed technique solves this issue.
Relative to the calibration methodologies that have been recently reported by our group, the algorithm in this article shows significant advantages [
30,
31]. In contrast to the approach in Ref. [
30], the feedback architecture is employed to approximate the parameters to the actual values, despite inherent stability concerns. By utilizing a feedforward calibration structure, this paper systematically enhances the stability and improves the entire performance of the calibration process. Relative to the framework in Ref. [
31], the proposed structure is characterized by its simplicity, low computational complexity, rapid convergence, and superior performance. The calibration structures in Refs. [
30,
31] present limited efficacy in the multi-frequency signals calibration, therefore imposing significant constraints on their practical applications. In contrast, the calibration architecture in this paper demonstrates the superior effectiveness in the context of the multi-frequency signals calibration.
Although the inclusion of the reference channel is a notable drawback in comparison to the structure without one, it has been demonstrated to be a strong-expansible and straight structure. The application of the method from Ref. [
2] to systems with more than 8 channels generates an exponential increase in hardware resource consumption at both the estimation module for derivative poly-phase filters and the matrix processing unit. However, the proposed structure in this paper can be expanded to accommodate additional channels without incurring any extra hardware consumption within the estimation module.
Given all that, this paper sufficiently validates the effectiveness of the algorithm by the co-simulation of the MATLAB and off-chip measured data. Compared with other literature [
9,
28,
29], the all-digital calibration framework in this paper demonstrates the notable advantages in terms of convergence speed, SFDR, SNR, hardware resource efficiency, and multi-frequency signals calibration. However, the approaches involve additional channels, necessitating a more complex layout, especially concerning the clock tree. In the context of the future endeavors, the forthcoming work will involve the refinement of the existing algorithmic framework, with a particular emphasis on optimizing both the area utilization and the performance. Our calibration methodology is slated for the implementation leveraging an ASIC library.