1. Introduction
The characterization of magnetic fields is fundamental in many relevant technological sectors, namely, automotive, transport, aero spatial, etc., [
1,
2]. In this scenario, different magnetic sensing principles have been proposed for the development of suitable magnetic sensors, such as giant magneto-resistive sensors [
3,
4], resonant coil sensors [
5], fluxgate sensors [
6,
7], Hall sensors [
8], spin valve sensors [
9], superconducting quantum interference devices (SQUID) [
10], etc. Among them, sensors based on the giant magneto-impedance effect (GMI) have been revealed as a powerful tool due to their faster response, smaller size, higher stability, lower cost, and especially their generally higher sensitivity during detection [
2,
11]. These features have enabled their use to spread in different fields, employing geometries and configurations such as amorphous wires, [
12] ribbons [
13], microwires [
14], or thin films [
15,
16], among others.
The GMI effect can be defined as the huge changes experienced by the high-frequency impedance,
Z = R + jX (where
R is the resistance and
X the reactance) of a ferromagnetic conductor when it is exposed to a static DC magnetic field
H [
2,
11]. The development of magnetic sensors based on the GMI effect has been gaining importance due to the growing necessity to tackle the detection of increasingly lower variations in the magnetic field value. This research effort demands the optimization of the device response during the detection procedure, a goal that can be accomplished, among other ways, through the enhancement of the synergetic coupling between the GMI sensing element and the signal conditioning circuitry. This circumstance turns the lock-in amplifier into one of the most suitable choices for signal detection in GMI-based applications.
A lock-in amplifier (LIA) is an electronic system used to recover very small AC signals in extremely noisy environments, even with noise and interference levels thousands of times larger than the amplitude of the target signal [
17,
18]. LIAs are also employed for precision measurement in scenarios with a good signal-to-noise ratio, but where extremely small amplitudes and/or phase variations need to be detected [
17]. The core of the LIA is based on a phase-sensitive detector (PSD), conceived for detecting the amplitude and phase variations in a signal at a given frequency previously known. The LIA applications include a wide range of disciplines in applied science and technology, such as spectroscopy, biomedical measurements, complex impedance characterization, and photometry to name just a few [
17]. During the last decades, the popularization of microprocessor and FPGA-based systems have led to the gradual substitution of traditional analog LIAs by digital implementations [
4,
18,
19,
20,
21], mainly due to their robustness and programmability without compromising their key specifications at the expense of higher complexity and extra computing power. This digitalization tendency is induced, in part, by the difficulty of implementing high-performance PSD circuits based on analog multiplication.
Analog multipliers are prone to generate distortion and DC offset due to their nonlinear operation [
22,
23,
24]. The magnitude of this offset increases with input amplitude and operation frequency, potentially compromising the proper operation of the system. More concretely, output DC offsets may be a severe problem in lock-in circuits based on analog multipliers, since the offset will be indistinguishable from the useful DC value provided by the PSD. Although several techniques for offset cancelation in multipliers have been proposed in the literature [
25,
26], they are conceived for canceling not only offset but also any DC component at the output, which makes them unsuitable for PSD applications. A different approach for the DC stabilization of analog multipliers using chopper techniques has been presented in [
24]. Although it preserves the DC input components, any useful information associated with the multiplier DC output will be lost as well. Hence, there is no effective solution for separating the useful DC value at the PSD output from the offset introduced by the analog multiplier, so its use in LIA design is limited to favorable conditions in which the generated offset is negligible, i.e., low operation frequencies. For instance, an analog LIA for phase measurement based on a PSD using the AD633 multiplier (Analog Devices) [
27] has been presented in [
28]. It provides high-resolution phase detection operating at 77 Hz.
An alternative approach consists of implementing the multiplication function by means of switched multiplier schemes [
7,
29,
30]. In this case, the sinusoidal reference signal is substituted by a ±1 square waveform of the same frequency. However, this PSD scheme leads to additional drawbacks: on the one hand, odd harmonics associated with the square waveform are introduced on the reference channel, which may induce misleading results if spectral components at these frequencies are present at the PSD input; on the other hand, the switched circuit may produce delay and jitter which may provoke synchronization problems at high frequencies [
31,
32].
As can be concluded from the above, the severity of both drawbacks linked to analog PSD design is strongly alleviated as the operation frequency decreases. In sensor-based applications, this frequency of operation is typically constrained by the sensor excitation frequency range. In particular, GMI amorphous wires and ribbons typically operate in the range of hundreds of kHz to a few MHz to maximize the GMI effect [
11], while other GMI sensing elements such as amorphous microwires or thin films generally operate concretely at larger frequencies, from several tens of MHz to even GHz values [
14,
15,
16]. In these latter cases, most of the related works in the literature employ expensive high-performance laboratory or computer-aided equipment for experimental measurements [
14,
15,
16], due to the difficulty of designing low-cost, compact, and portable analog electronics at these frequencies. However, the operation frequency of GMI wires is suitable for the implementation of simple ad hoc lock-in systems based on analog PSDs, but, in contrast, the aforementioned problems associated with analog multiplication may begin to be significant when operating at hundreds of kHz.
In this context, a subsampling technique for frequency downscaling in analog LIA design using GMI sensors is proposed in this work. Specifically, both the output sensor and reference signals at the PSD inputs, initially at 200 kHz, are frequency-downscaled to 1 kHz to exploit the inherent advantages of signal multiplication at low frequencies. In this way, the functionality of the PSD remains unaffected if both input signals are scaled by the same factor, while the accuracy of the LIA is strongly enhanced. As proved later, the benefits of the proposed technique are evident both for conventional analog and switched multipliers. This frequency downscaling process is carried out by utilizing a subsampling process [
33,
34] employing sample-and-hold (SH) circuits [
35].
Although the proposed technique has been applied to GMI sensors in this work, it is theoretically extensible to any LIA using an analog PSD. A practical limitation may arise from the growing complexity of implementing a precise SH block as the operating frequency increases. However, accurate SH circuits can be easily assembled using discrete components for working at frequencies of hundreds of kHz.
The paper is organized as follows: a brief review of the LIA fundamentals is provided in
Section 2. In
Section 3, the proposed subsampling technique for frequency downscaling in analog LIAs is presented. The complete experimental setup including the GMI wire, sensor interface, laboratory instrumentation, and the assembled LIA circuits is explained in
Section 4. The experimental results are presented and discussed in
Section 5, and finally, some conclusions are drawn in
Section 6.
2. Fundamentals of LIAs
An LIA can be understood as a measurement system for signal recovery operating on the PSD principle, also supported by amplification and processing stages [
17]. The LIA approach requires, by definition, some previous knowledge about the signal that must be detected, and the parameters of the reference signal must be adjusted accordingly. The principle of operation of a PSD is shown in
Figure 1a.
Let’s assume that the input signal
Vin is formed by the sum of the signal of interest
Ai·sin(
ωi·t +
ϕi) and a generic signal
n(
t) that contains unwanted components at other frequencies. Using a sinusoidal reference signal
Vref1 = Ar·sin(
ωr·t + ϕr), whose frequency is precisely synchronized with the target signal (
ωr =
ωi), the multiplier output
Vmult1 will be given by
Hence, the multiplier performs a synchronous detection that leads to an output voltage with a DC component corresponding to the first term of (1). This DC term can be straightforwardly obtained by using a very low-frequency low-pass filter (LPF), resulting in:
Note from (2) that amplitude and phase variations cannot be detected at the same time by the PSD of
Figure 1. On the one hand, variations in
ϕi can be precisely measured if the amplitude of the input signal A
i remains constant. In this case, the PSD output voltage will be proportional to cos(
ϕr −
ϕi). On the other hand, amplitude variations can be faithfully detected if a zero-phase difference between
Vref1 and the signal of interest is forced, i.e.,
ϕr =
ϕi. The implementation of the equivalent PSD employing switched multipliers is illustrated in
Figure 1b. The sinusoidal reference signal
Vref1 has been substituted by a ±1 square wave of the same frequency (
Vref2), whose Fourier series is given by
In this case, assuming again that
ωr =
ωi, the following multiplier output
Vmult2 is obtained:
Despite the complexity of (4), note that the final low-pass filter of the multiplier of
Figure 1b will remove most of the terms as occurred with (2), just keeping the first DC term:
Although Equation (5) is ideally similar to Equation (2), it must be considered that if n(t) contains spectral components at odd multiple frequencies of
they will be multiplied by the corresponding harmonic of the square reference signal according to the last term of Equation (4). Hence, undesired DC terms will be added to Equation (5) and the accuracy of the PSD will be degraded. Moreover, another limitation may arise from the potential delay, t
d, that the switch SW of
Figure 1b may introduce. If this delay, t
d, is low enough to become negligible against the period of the PSD input signal (t
d << 2π/
), then Equation (5) will remain unaffected in practice. However, if t
d provokes a significant phase-shifting on the reference signal, the effective value of
will be modified, leading to a source of error in
VDCout2.Once the PSD fundamentals have been addressed, a basic LIA architecture for amplitude detection is shown in
Figure 2a. It operates as follows: a device-under-test is excited by an input sinusoidal signal
Vi. Although in some applications the signal of interest might be related to higher harmonics or intermodulation products [
36,
37], its own fundamental frequency is targeted in most of the cases, so the reference signal V
ref can be directly generated from the input signal. A phase-shifting block is included to ensure a phase matching between both PSD inputs so that the final DC output,
, is proportional to the amplitude of the signal of interest. Specifically,
where
ADUT is the amplitude at the frequency of interest at the device-under-test output,
Ai is the amplitude of the excitation signal, and
G is a generic gain introduced by the filter, multiplier or amplifier blocks throughout the system.
An alternative approach providing simultaneous detection of amplitude and phase variations is shown in
Figure 2b. This circuit incorporates two PSDs operating with in-phase and quadrature reference signals, so the corresponding output voltages
VIrec and
VQrec can be understood as the cartesian representation of the recovered signal
Vrec. Therefore, its instantaneous amplitude and phase can be straightforwardly calculated as:
The performance of practical lock-in systems may also be limited by different tradeoffs between the parameters of the LIA, such as dynamic range, linearity, dynamic reserve, or output stability [
17], which must be adapted to each final application.
3. Proposed LIAs Using Subsampling for Frequency Downscaling
The proposed frequency downscaling technique based on subsampling and using SH circuits is explained in this Section. It has been conceived in general for analog LIA design, with the purpose of alleviating the aforementioned drawbacks associated with the PSD design at moderate and high frequencies. Nevertheless, the implementation presented in this work is focused on GMI sensor applications operating at frequencies of hundreds of kHz. Firstly, the linearity of analog multipliers is enhanced as the operation frequency decreases [
22,
32], while the output offset is strongly reduced. Secondly, the delay and jitter introduced by switched multipliers on the reference channel become negligible if the period of the signal of interest becomes large enough [
31,
32].
On this basis, an SH block has been included in the LIA topologies presented in
Figure 3 for frequency reduction by means of a subsampling process. Both circuits are based on the general schemes of
Figure 2 and have been adapted for detecting the fundamental harmonic at the output of a GMI sensor interface operating at 200 kHz. The PSD of the LIA in
Figure 3a is based on the switched multiplication principle. In this case, the reference square wave is extracted from the sensor output signal itself through a comparator, leading to a simplified version of the scheme in
Figure 2a. Specifically, the phase-shifting block has been excluded to avoid complicating the system beyond our purpose of validating the benefits of the proposed technique. Furthermore, the circuit of
Figure 3b is a straightforward implementation of the scheme in
Figure 2b using analog multipliers. Note that the SH blocks are properly placed for down-converting the operation frequency at both PSD inputs. A sampling frequency
fs = 199 kHz has been employed so that the subsampled output signal is scaled down to
fin −
fs = 1 kHz. The fundamental of this subsampling process is explained in detail below.
3.1. Fundamentals of Subsampling
Subsampling can be defined as the sampling process of a bandpass signal using a frequency below the Nyquist frequency. Considering that the signal bandwidth (BW) is generally much lower than the center frequency
fc, the subsampling may be carried out without aliasing between spectrum replicas [
34]. Specifically, the spectrum of a sampled signal is given by
where
X(
f) is the spectrum of the original signal, and
fs and
Ts are, respectively, the sampling frequency and period. As explained in detail in [
33], when a bandpass signal is subsampled at a frequency
fs <
fc, spectrum replicas are generated at −
m·
fs +
fc, while mirrored replicas appear at (
m + 1)·
fs −
fc, where
m is an integer. A low-frequency replica of the original signal will be obtained at
This signal replica at
fL can be extracted without aliasing if the following condition holds [
33,
38]:
where
nr is the number of replicas of the spectrum of the signal between 0 and
fc-(
BW/2). Note that at this point, the circuits in
Figure 3 employ a sinusoidal excitation signal whose bandwidth ideally tends to zero, so the constraints stated in Equation (11) are relaxed in this application. As indicated in
Figure 3, a frequency
fs = 199 kHz has been chosen for sampling the signal at the sensor excitation frequency (200 kHz), so a low-frequency replica at
fL =1 kHz would be obtained according to Equation (10).
To provide some insight, an illustrative example of the subsampling of a sinusoidal signal with a generic frequency
fc is depicted in
Figure 4. The spectrum of the original sinewave is plotted in
Figure 4a, assuming the presence of second and third harmonics. A sampling frequency
fs slightly lower than
fc has been established. According to Equation (9), the spectrum shown in
Figure 4b is obtained for the ideally subsampled signal. A low-frequency replica of the original signal is obtained at
fL = fc −
fs, as stated in Equation (10). Note also that replicas of the second and third harmonics appear at frequencies 2·
fL and 3·
fL, respectively, so a whole downscaling of the spectrum in
Figure 4a has been obtained. It is worth mentioning that a subsampling frequency of
fs = 4·
fc/nr (with
nr an odd natural number) is recommended in communication systems for obtaining a signal replica at
fs/4, thus relaxing the subsequent filter requirements for signal recovery [
33,
38]. However, a different criterion has been employed in the current application, since the benefits of the proposed downscaling technique enhance as
fL decreases. Hence, the lower limit of
fL is uniquely determined in practice by the electronics’ flicker noise, which may become dominant at frequencies below the kHz range [
18], and by the possible requirements in terms of measurement speed, which depends on the target application.
3.2. Sample-and-Hold Considerations
The previous analysis based on Equation (9) assumes that an ideal sampling is performed, i.e., the original signal is multiplied by a unit impulse train. However, when the basic SH architecture in
Figure 5 is employed in practice, the original signal is multiplied by a train of rectangular pulses, as illustrated in
Figure 6. The system operates as follows: an input amplifier buffers the input signal (
Figure 6a) to charge the capacitor C
HOLD. During the track mode (switch SW closed), the capacitor voltage follows the input signal, while in the hold mode (switch open) the sampled voltage is retained in C
HOLD. A second amplifier with a very high input impedance is employed for driving the C
HOLD voltage, as well as preventing the capacitor from discharging prematurely [
35]. As a result, a pulse-amplitude-modulated (PAM) signal with a pulse duration equal to the sampling period T
S is obtained [
39], as depicted in
Figure 6b. Under this assumption, Equation (9) becomes
where
H(
f) is the Fourier transform of the rectangular pulse shape, in turn, given by
Returning to the example of
Figure 4, the spectrum in
Figure 4c corresponds to the subsampling process of the original signal using rectangular pulses with a duration of
Ts. Note from Equation (12) that the multiplication in the frequency domain of an ideally sampled signal by a sinc shape will distort the original spectrum proportionally to its bandwidth. However, in the particular case of a sinusoidal signal, the sinc multiplication just introduces an attenuation
Ts·|
sinc(
fL·
Ts)| at the fundamental frequency, and even the relative magnitude ratio with respect to the second and third harmonics (further attenuated) is increased, as can be observed in
Figure 4c. Anyway, both effects become negligible if the ratio
fc/
fL is high enough.
5. Measurement Results
The voltage V
sens at the output of the sensor interface in
Figure 7a was monitored by the digital oscilloscope. The magnitude of the fundamental harmonic was accurately calculated and averaged using the FFT function of the device software, while the phase variations were extracted from the time domain signal employing the same device. Both parameters were taken as references for evaluating the precision of the proposed LIAs. The LIA topology of
Figure 3 (implemented according to
Figure 7) was tested with and without the SH blocks, i.e., using frequency downscaling at the PSD inputs (to 1 kHz), and with the conventional PSD (at 200 kHz). The distance between the neodymium magnet and the GMI sensor was varied from 50 mm to 12.5 mm in steps of 2.5 mm.
The measurement results of the amplitude of the fundamental harmonic of V
sens (see
Figure 7a) are plotted in
Figure 9. In
Figure 9a, the black solid line corresponds to the reference value of V
sens provided by the FFT function at 200 kHz, while the outputs of both LIAs using frequency downscaling are drawn in blue (single-output LIA) and red (two-output LIA) dashed lines, respectively. Note that the curves of both LIAs have been plotted with a scale factor such that the voltage swing between maximum and minimum values is the same as for the reference curve. In this way, a precise comparison in terms of offset can be performed. The same procedure is used in
Figure 9b for the outputs of the conventional LIAs without frequency reduction. To provide more insight, the LIAs measurement error with respect to the FFT reference values is shown in
Figure 9c for all the previous cases. Regarding the two-output LIA, the signal magnitude was externally calculated from the two captured output voltages (V
Iout and V
Qout) according to Equation (7). Note that the error introduced by the analog multipliers was considerably reduced when frequency-downscaled input signals were employed. Specifically, the mean voltage deviation throughout the experiment decreased from 628.1 µV to 74.9 µV, which corresponded to a reduction factor of 8.39. The benefits of the proposed technique are even more noticeable in the case of the single-output LIA, in which the delay of the switches seriously deteriorated the accuracy of the PSD when the conventional topology was used. In this case, the effect of the switch delay was strongly alleviated thanks to the frequency downscaling, reducing the mean deviation of the voltage V
SMout in
Figure 7a by a factor of 21.15, i.e., from 3.743 mV to 177 µV.
Furthermore, the effect of the proposed technique on the phase response of the two-output LIAs (scheme in
Figure 3b and outputs V
Iout and V
Qout in
Figure 7a) were studied. In this case, a specific point needed to be established as a phase reference value (0°), so phase variations could be measured and compared. A sensor position 50 mm away from the magnet was arbitrarily set as the reference. From this point, the phase variation was monitored as the magnet and sensor approach, leading to the results plotted in
Figure 10a. The reference black solid line was obtained by the phase function of the digital oscilloscope, which calculated the phase difference between V
sens in
Figure 7a and the input signal V
in, which served as a stable reference sinewave. A digital bandpass filtering centered at 200 kHz was also performed over both signals in order to attenuate possible higher-order harmonics that might have degraded the measurement accuracy. On the other hand, the blue and red dashed lines were obtained by applying Equation (8) to the LIA output voltages (calculations are performed externally), with and without employing the proposed technique. The representation of the phase errors is provided in
Figure 10b.
The benefits of the proposed technique concerning phase measurement accuracy are not as significant as for the magnitude, as can be seen in
Figure 10. This result was expected since if the same DC offset is added to both LIA output voltages, V
Iout and V
Qout, the error in the tan
−1(V
Qout/V
Iout) operation may be partially compensated. In fact, this error will be ideally zero if both in-phase and quadrature components have the same value initially.
Nevertheless, the proposed technique also had a positive effect on the precision of the phase variations measurement. Note that the mean errors obtained from both lines in
Figure 10b were conditioned by the arbitrary choice of the reference position (phase = 0°). For the sake of reliability, the mean phase deviation was calculated N times for N different reference points from the data in
Figure 10, where N = 16 is the total number of points for each curve. These N mean values were subsequently averaged, leading to final mean error values of 0.130° for the conventional circuit and 0.097° for the circuit using frequency downscaling. Hence, the proposed technique reduced the phase measurement error by >25% compared to the conventional lock-in topology.