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REVIEW 3 major objections 5 minor 46 references

Practical Modulo Sampling: Mitigating High-Frequency Components

T0 review · 3 major / 5 minor · reviewed 2026-08-10 · deepseek-v4-flash

Pith's one-line read Multiplying the folded signal by a delta comb before low-pass filtering lets a realistic ADC produce exactly the samples of an ideal modulo sampler.

desk verdict A sound and useful adaptation of MWC to modulo sampling, backed by a real hardware prototype, but the load-bearing error from the finite comb is defined and never bounded, so the practical 'equivalent to an ideal ADC' claim outruns the evidence. read the letter →

arxiv 2501.11330 v1 pith:JWTEBBYI submitted 2025-01-20 eess.SP

classification eess.SP
keywords modulosamplingdynamicrangeanalog-to-digitalconverterhigh-frequencycomponentslow-passfiltercombgeneratorquantizationnoisehardwareprototype
verification ladder T0 review T1 audit T2 compute T3 formal

The pith

A machine-rendered reading of the paper's core claim, the machinery that carries it, and where it could break.

The reading

Modulo (folding) sampling lets an ADC capture signals much larger than its dynamic range, but the modulo operation creates high-frequency components that ordinary ADCs filter out, corrupting the samples. This paper proposes an analog mixer plus low-pass filter ahead of a realistic ADC: multiply the folded signal by a periodic comb, filter, and sample. Theorem 1 proves that the resulting samples equal the ideal pointwise samples of the folded signal, so any existing modulo recovery method applies unchanged. Simulations with 6-bit and 8-bit quantizers and a hardware prototype show that the comb-based pipeline tracks the ideal modulo sampler and beats classical sampling at oversampling rates of at least 5. The paper's central claim is that realistic, low-bandwidth ADCs can therefore replace the high-specification ADCs that modulo sampling previously required.

What carries the argument

The load-bearing object is the delta-comb multiplier $p(t)=\sum_{n\in\mathbb{Z}} \delta(t-nT_s)$ placed between the analog modulo operator and the low-pass filter. Multiplying by the comb periodizes the folded signal's spectrum with spacing $2\pi/T_s$; the low-pass filter with cutoff $\pi/T_s$ then selects the baseband copy, and Poisson's summation formula equates the DTFT of the samples $y[nT_s]$ with the DTFT of the ideal modulo samples $M_\lambda x[nT_s]$. This identity is what lets a narrowband ADC emulate a wideband one. In the hardware realization the infinite comb is approximated by a finite Fourier sum with $N=2000$ harmonics, and the resulting mismatch is tracked as $E_{\text{mod-HF}}$ in the total error budget $E_{\text{mod}}=E_{\text{mod-HF}}+E_{\text{mod-Q}}$.

What would settle it

Measure $E\left[|y[nT_s]-M_\lambda x[nT_s]|^2\right]$ for the finite-comb system with $N=2000$ while increasing the input signal's maximum frequency or lowering the oversampling ratio at fixed bit depth; if this high-frequency error grows to dominate the quantization error and the recovered signal degrades well below the ideal-sampler baseline, the claimed practical equivalence fails.

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Extended reading notes

Core claim

The paper's central discovery is an exact equivalence: if the folded signal $M_\lambda x(t)$ is multiplied by a delta comb $p(t)=\sum\delta(t-nT_s)$ and then passed through a low-pass filter with cutoff $\pi/T_s$ before sampling at rate $T_s$, the output samples satisfy $y[nT_s]=M_\lambda x[nT_s]$. The proof uses the fact that multiplying by the comb periodizes the spectrum of the folded signal with period $2\pi/T_s$; Poisson's formula identifies the DTFT of the desired samples with this periodized spectrum, and the filter keeps exactly the baseband copy. Consequently, the samples from a realistic ADC are identical to those an ideal wideband pointwise sampler would produce from the folded signal. The paper then decomposes the practical error into $E_{\text{mod-HF}}$, caused by replacing the infinite comb with a finite Fourier sum of $N=2000$ terms, plus the quantization error $E_{\text{mod-Q}}$, and shows experimentally that the comb brings $E_{\text{mod-HF}}$ below the quantization error. A hardware prototype built from a step-recovery-diode comb generator, an analog multiplier, and a 25 kHz low-pass filter confirms the sample alignment and supports recovery with a 1-bit side-information algorithm.

Load-bearing premise

The practical equivalence rests on the unquantified assumption that replacing the delta comb by a finite Fourier sum with $N=2000$ harmonics makes $E_{\text{mod-HF}}$ negligible; the paper supports this only with empirical curves in Figures 13 and 14, not with a theoretical bound.

Editorial extensions

If this is right

  • Any existing modulo recovery algorithm for bandlimited signals, including quantization-aware ones, can be applied directly to the samples from a realistic ADC.
  • The ADC's bandwidth requirement drops back to the input signal's Nyquist rate; the analog mixer and filter absorb the high-frequency burden.
  • At oversampling rates of at least 5 and with 6- or 8-bit quantizers, the proposed pipeline approaches ideal modulo sampling and outperforms a classical infinite-dynamic-range ADC.
  • Because $E_{\text{mod-HF}}$ and quantization error are independent, improving the comb's harmonic content is a direct route to better high-precision recovery.
  • A hardware prototype validates the theoretical equivalence, making unlimited-dynamic-range sampling viable with off-the-shelf components.

Reading between the lines

Editorial extensions of the paper, not claims the author makes directly.

  • The paper does not provide a theoretical bound on $E_{\text{mod-HF}}$ as a function of $N$, signal bandwidth, and oversampling ratio; deriving one would turn the empirical choice $N=2000$ into a design rule and is a natural next step.
  • The same comb-then-filter identity only relies on periodization and baseband selection, so it likely carries over to shift-invariant and finite-rate-of-innovation modulo recovery, where the same high-frequency problem arises.
  • A simpler periodic waveform, such as a sine wave, could replace the full comb at the price of weighting the folded spectrum; the trade-off between hardware simplicity and spectral content is not explored in the paper.
  • Real low-pass filters have passband ripple and finite stopband rejection, so quantifying how filter nonideality translates into $E_{\text{mod-HF}}$ is a testable extension of the hardware results.
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Editorial analysis

A structured set of objections, weighed in public.

Desk editor's note, referee report, and a circularity audit.

Referee Report

3 major / 5 minor

Summary. The manuscript proposes and prototypes a hardware front-end for modulo sampling in which the folded signal is multiplied by a periodic comb and low-pass filtered before sampling, so that a narrowband ADC can capture samples claimed to equal those of an ideal wideband modulo sampler. Theorem 1 proves this equality for an ideal delta comb and an ideal LPF. The paper then describes an SRD-based comb generator, an AD834 mixer, and a 7th-order Butterworth LPF, and reports simulations and hardware experiments comparing the proposed approach with ideal modulo sampling, direct LPF-modulo sampling, and classical sampling.

Significance. If the ideal equivalence were robust to finite comb length and nonideal filtering, the paper would be a genuine practical step toward modulo ADCs built from standard components. The clean ideal-theoretic result in Theorem 1 is correct and useful, and the hardware effort is concrete, with a component-level prototype and reproducible-looking experiments. The strength of the paper is therefore the combination of a valid sampling-theoretic identity with a real hardware demonstration. However, the central practical claim that the samples are 'equivalent' to an ideal high-spec ADC is not established, because the finite-comb error E_mod-HF is only defined and never bounded, and the nonideal filter is not analyzed.

major comments (3)
  1. [Section II-C, Eq. (10)] The statement that the practical system yields samples equivalent to those of an ideal high-spec ADC is not established, because E_mod-HF is only defined and never bounded. Theorem 1 applies to p(t) = Σ_n δ(t − nT_s), while the practical system uses the finite Fourier sum p_N(t) = Σ_{k=−N}^{N} e^{j2πkt/T_s}. Since M_λx is not bandlimited, the output differs from M_λx[nT_s] by a term that depends on the high-frequency tail of M_λx and on the missing harmonic copies with |k| > N. The paper provides no upper bound on E_mod-HF as a function of N, T_s, λ, the input bandwidth, or the oversampling ratio, and no condition ensuring E_mod-HF is below E_mod-Q. Figures 13 and 14 show a single simulation ensemble at N=2000 with no error bars or worst-case analysis. This is load-bearing, because the abstract's 'equivalent' claim and the statement that 'any existing modulo recovery method can be applied effectively' depend on this unquantified numerical choice.
  2. [Section III-D and Fig. 11] The prototype uses a 7th-order Butterworth LPF with a −3 dB cutoff at 25 kHz, whereas Theorem 1 assumes an ideal LPF with cutoff π/T_s. The nonideal filter's passband ripple and finite stopband attenuation change y[nT_s] relative to the ideal value, and this deviation is not modeled in E_mod-HF. Moreover, the comb generator's output harmonic amplitudes are not reported, so it is not verified that the hardware implements the equal-amplitude Fourier sum assumed in Section II-C. The paper should quantify the filter-induced error and the comb spectral error, or provide design criteria such as required stopband attenuation and harmonic flatness in terms of the target MSE.
  3. [Section IV-B versus Section III-A] There is an internal inconsistency in the hardware description. Section III-A defines Sampler 3 as classical sampling of the input and Sampler 4 as the folded signal passed through a 25 kHz LPF, but Section IV-B says Sampler 3 records the folded signal after passing through a LPF, while the caption of Fig. 15 labels the LPF path as Sampler 4. Additionally, Section IV-B states that the hardware recovery used 'straightforward unwrapping' [45], [46], not the quantization-robust algorithm of [33] that Section II-C introduces and the abstract credits for the results. The sampler labeling should be corrected, and the text should clarify which recovery algorithm was used in each hardware experiment, because the current description makes the hardware validation difficult to interpret.
minor comments (5)
  1. [Theorem 1 statement] The statement contains a typo: 'than y[nT_s]' should be 'then y[nT_s]'.
  2. [Section II-C] The finite-comb definition writes p(t) = Σ_{k=−N}^{N} e^{jk/T_s t}; the exponent is missing the factor 2π and should read e^{j2πkt/T_s}.
  3. [Section IV-A] The text says that the inclusion of the comb generator 'significantly reduces E_mod-Q', but E_mod-Q is the quantization error and is independent of the comb; the sentence should refer to E_mod-HF.
  4. [Equation (10)] The claim that the errors are independent because quantization occurs after analog processing is not by itself a justification for adding the two MSE terms; independence of the error processes should be stated or derived, and the MSE should be defined over the relevant randomness.
  5. [General] There are minor grammatical errors throughout, such as 'The theorem above provide' and 'which shown in the figure'; a careful proofreading pass would improve clarity.

Circularity Check

0 steps flagged · score 0.0 of 10

No significant circularity: the central equivalence (Theorem 1) is derived from Fourier/Poisson first principles, and the finite-comb approximation is explicitly acknowledged as an error term rather than disguised as a prediction.

full rationale

The paper's central claim is Theorem 1, which states that with an ideal delta-comb p(t), the mixer+LPF front end produces samples y[nTs] identical to the ideal modulo samples M_lambda x[nTs]. The proof is a direct derivation from the CTFT/DTFT definitions and Poisson's summation formula; it does not assume the conclusion. The practical implementation replaces the ideal comb by a finite Fourier sum with N=2000, and the paper explicitly defines the resulting residual as E_mod-HF in Eq. (10), adding it to the quantization error instead of claiming it is zero. This is an acknowledged approximation and a validation gap, not a circular step. The many self-citations (e.g., [28]–[32]) are used as background, as recovery algorithms with independent guarantees, or as hardware components; no load-bearing claim is justified solely by citing the same authors. The lack of a theoretical bound on E_mod-HF as a function of N, bandwidth, and oversampling ratio is a correctness/evidence concern, not a circularity, because the equivalence was not fitted from the empirical curves in Figures 13 and 14. Accordingly, the derivation chain is self-contained and no circular step is identified.

Assumptions & free parameters 1 free parameters · 5 assumptions · 0 invented entities

The central claim relies on standard Fourier analysis, the bandlimited signal model, and ideal components in the proof. The only hand-set parameter in the practical implementation is the finite comb length N=2000, whose induced error is not quantified. No new physical entities are postulated.

free parameters (1)
  • Number of comb harmonics N = 2000
    The ideal delta comb in Theorem 1 is replaced in Section II-C by a finite Fourier sum with N=2000 terms. The resulting error E_mod-HF is acknowledged but no optimal N, bound, or fitting procedure is given; it is a hand-chosen implementation parameter.
assumptions (5)
  • domain assumption The input signal x(t) is bandlimited with finite Nyquist period T
    Required for the modulo recovery results and for the Poisson summation argument; stated in Section II-A.
  • ad hoc to paper The low-pass filter is ideal with cutoff frequency pi/T_s
    Theorem 1 assumes an ideal brick-wall LPF; the hardware uses a 7th order Butterworth, so the equivalence is approximate.
  • standard math The delta comb p(t) = sum_n delta(t - nT_s) is a valid distribution and multiplication with M_lambda x is well-defined
    The proof uses distributional Fourier transforms; the signal is assumed to have finite norm, so the products are formal.
  • standard math Poisson summation formula holds for M_lambda x
    Equation (3) uses Poisson's formula to relate the DTFT of samples to the CTFT of the folded signal.
  • domain assumption The analog multiplier is ideal (no bandwidth limitation, no non-linearity)
    The hardware uses AD834; its finite bandwidth and non-ideal behavior are not modeled in the proof.

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Pith. "Pith review of Practical Modulo Sampling: Mitigating High-Frequency Components." pith.science (2026). https://pith.science/paper/JWTEBBYI

@misc{pith2026250111330,
  author       = {Pith},
  title        = {Pith review of: Practical Modulo Sampling: Mitigating High-Frequency Components},
  year         = {2026},
  howpublished = {\url{https://pith.science/paper/JWTEBBYI}},
  note         = {Machine review of arXiv:2501.11330}
}
read the original abstract

Recovering signals within limited dynamic range (DR) constraints remains a central challenge for analog-to-digital converters (ADCs). To prevent data loss, an ADCs DR typically must exceed that of the input signal. Modulo sampling has recently gained attention as a promising approach for addressing DR limitations across various signal classes. However, existing methods often rely on ideal ADCs capable of capturing the high frequencies introduced by the modulo operator, which is impractical in real-world hardware applications. This paper introduces an innovative hardware-based sampling approach that addresses these high-frequency components using an analog mixer followed by a Low-Pass Filter (LPF). This allows the use of realistic ADCs, which do not need to handle frequencies beyond the intended sampling rate. Our method eliminates the requirement for high-specification ADCs and demonstrates that the resulting samples are equivalent to those from an ideal high-spec ADC. Consequently, any existing modulo recovery algorithm can be applied effectively. We present a practical hardware prototype of this approach, validated through both simulations and hardware recovery experiments. Using a recovery method designed to handle quantization noise, we show that our approach effectively manages high-frequency artifacts, enabling reliable modulo recovery with realistic ADCs. These findings confirm that our hardware solution not only outperforms conventional methods in high-precision settings but also demonstrates significant real-world applicability.

Figures

Figures reproduced from arXiv: 2501.11330 by the authors.

Figure 1
Figure 1. (a) BL signal alongside its clipped version, sampled with an ADC [PITH_FULL_IMAGE:figures/full_fig_p001_1.png] view at source ↗
Figure 2
Figure 2. Illustration of the modulo recovery framework for BL signals as [PITH_FULL_IMAGE:figures/full_fig_p003_2.png] view at source ↗
Figure 3
Figure 3. The proposed block diagram for sampling and modulo recovery using [PITH_FULL_IMAGE:figures/full_fig_p004_3.png] view at source ↗
Figures from the paper (11 more)
Figure 5
Figure 5. Figure 5: Realistic Modulo hardware board with highlighted components shown from two perspectives: the top image highlights key components such as the [PITH_FULL_IMAGE:figures/full_fig_p005_5.png]
Figure 4
Figure 4. Figure 4: Sampler 2 (TEENSY-S1) receives the modulo signal [PITH_FULL_IMAGE:figures/full_fig_p005_4.png]
Figure 6
Figure 6. Figure 6: Four analog input signals with four synchronized 50KHz ADC [PITH_FULL_IMAGE:figures/full_fig_p006_6.png]
Figure 7
Figure 7. Figure 7: Multiplied signal with COMB Generator at 50KHz steps, passed [PITH_FULL_IMAGE:figures/full_fig_p006_7.png]
Figure 8
Figure 8. Figure 8: COMB Generator with 50KHz steps and SRD. [PITH_FULL_IMAGE:figures/full_fig_p006_8.png]
Figure 12
Figure 12. Figure 12: Oscilloscope snapshot: yellow—input BL signal, green—folded [PITH_FULL_IMAGE:figures/full_fig_p007_12.png]
Figure 11
Figure 11. Figure 11: Attenuation versus frequency LPF 25KHz S parameters. In the [PITH_FULL_IMAGE:figures/full_fig_p007_11.png]
Figure 13
Figure 13. Figure 13: Comparison of quantization errors in 6-bit (left) and 8-bit (right) systems without the comb generator. [PITH_FULL_IMAGE:figures/full_fig_p008_13.png]
Figure 14
Figure 14. Figure 14: Comparison of quantization errors in 6-bit (left) and 8-bit (right) systems incorporating the comb generator. [PITH_FULL_IMAGE:figures/full_fig_p009_14.png]
Figure 15
Figure 15. Figure 15: Comparison of sampled signals: Right shows the proposed realistic [PITH_FULL_IMAGE:figures/full_fig_p009_15.png]
Figure 16
Figure 16. Figure 16: Comparison of recovered signals: (a) Recovered BL signal using [PITH_FULL_IMAGE:figures/full_fig_p009_16.png]

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Reference graph

Works this paper leans on

46 extracted references · 45 canonical work pages

  1. [45]

    A new phase unwrapping algorithm,

    J. Tribolet, “A new phase unwrapping algorithm,” IEEE Transactions on Acoustics, Speech, and Signal Processing , vol. 25, no. 2, pp. 170–177, 1977

  2. [46]

    Satellite radar interferometry: Two-dimensional phase unwrapping,

    R. M. Goldstein, H. A. Zebker, and C. L. Werner, “Satellite radar interferometry: Two-dimensional phase unwrapping,” Radio science , vol. 23, no. 4, pp. 713–720, 1988

  3. [33]

    Modulo sampling with 1-bit side information: Performance guarantees in the presence of quantization,

    N. I. Bernardo, S. B. Shah, and Y . C. Eldar, “Modulo sampling with 1-bit side information: Performance guarantees in the presence of quantization,” in IEEE International Symposium on Information Theory (ISIT), July 2024

  4. [1]

    Y . C. Eldar, Sampling theory: Beyond bandlimited systems . Cambridge University Press, 2015

  5. [2]

    Sub-Nyquist sampling,

    M. Mishali and Y . C. Eldar, “Sub-Nyquist sampling,” IEEE Signal Processing Magazine, vol. 28, no. 6, pp. 98–124, 2011

  6. [3]

    Restoring lost samples from an oversampled band-limited signal,

    R. Marks, “Restoring lost samples from an oversampled band-limited signal,” IEEE transactions on acoustics, speech, and signal processing , vol. 31, no. 3, pp. 752–755, 1983

  7. [4]

    Error of linear estimation of lost samples in an oversampled band-limited signal,

    R. Marks and D. Radbel, “Error of linear estimation of lost samples in an oversampled band-limited signal,” IEEE transactions on acoustics, speech, and signal processing , vol. 32, no. 3, pp. 648–654, 1984

  8. [5]

    Restoring a clipped signal,

    J. S. Abel, “Restoring a clipped signal,” in Acoustics, Speech, and Signal Processing, IEEE International Conference On. IEEE Computer Society, 1991, pp. 1745–1748

Show all 46 references
  1. [6]

    Clip correction in wireless lan receivers,

    R. Rietman, J.-P. Linnartz, and E. P. de Vries, “Clip correction in wireless lan receivers,” in 2008 European Conference on Wireless Technology. IEEE, 2008, pp. 174–177

  2. [7]

    J. P. A. P ´erez, S. C. Pueyo, and B. C. L ´opez, Automatic gain control . Springer, 2011

  3. [8]

    A review of automatic gain control theory,

    D. Mercy, “A review of automatic gain control theory,” Radio and Electronic Engineer, vol. 11, no. 51, pp. 579–590, 1981

  4. [9]

    On the recovery of a band-limited signal, after instantaneous companding and subsequent band limiting,

    H. Landau, “On the recovery of a band-limited signal, after instantaneous companding and subsequent band limiting,” Bell System Technical Journal, vol. 39, no. 2, pp. 351–364, 1960

  5. [10]

    The recovery of distorted band-limited signals,

    H. Landau and W. Miranker, “The recovery of distorted band-limited signals,” Journal of Mathematical Analysis and Applications , vol. 2, no. 1, pp. 97–104, 1961

  6. [11]

    A wide dynamic-range cmos image sensor using self-reset technique,

    D. Park, J. Rhee, and Y . Joo, “A wide dynamic-range cmos image sensor using self-reset technique,” IEEE Electron Device Letters, vol. 28, no. 10, pp. 890–892, 2007

  7. [12]

    An implantable cmos image sensor with self-reset pixels for functional brain imaging,

    K. Sasagawa, T. Yamaguchi, M. Haruta, Y . Sunaga, H. Takehara, H. Takehara, T. Noda, T. Tokuda, and J. Ohta, “An implantable cmos image sensor with self-reset pixels for functional brain imaging,” IEEE Transactions on Electron Devices , vol. 63, no. 1, pp. 215–222, 2015

  8. [13]

    An activity-triggered 95.3 db dr −75.6 db thd cmos imaging sensor with digital calibration,

    J. Yuan, H. Y . Chan, S. W. Fung, and B. Liu, “An activity-triggered 95.3 db dr −75.6 db thd cmos imaging sensor with digital calibration,” IEEE journal of solid-state circuits , vol. 44, no. 10, pp. 2834–2843, 2009

  9. [14]

    Unlimited dynamic range analog-to-digital conversion,

    A. Krishna, S. Rudresh, V . Shaw, H. R. Sabbella, C. S. Seelamantula, and C. S. Thakur, “Unlimited dynamic range analog-to-digital conversion,” arXiv preprint arXiv:1911.09371 , 2019. 10

  10. [15]

    On unlimited sampling and reconstruction,

    A. Bhandari, F. Krahmer, and R. Raskar, “On unlimited sampling and reconstruction,” IEEE Transactions on Signal Processing , vol. 69, pp. 3827–3839, 2020

  11. [16]

    Above the Nyquist rate, modulo folding does not hurt,

    E. Romanov and O. Ordentlich, “Above the Nyquist rate, modulo folding does not hurt,” IEEE Signal Processing Letters, vol. 26, no. 8, pp. 1167– 1171, 2019

  12. [17]

    High dynamic range sensing using multi-channel modulo samplers,

    G. Lu and H. Liu, “High dynamic range sensing using multi-channel modulo samplers,” in 2020 IEEE 11th Sensor Array and Multichannel Signal Processing Workshop (SAM) . IEEE, 2020, pp. 1–5

  13. [18]

    Unlimited sampling from theory to practice: Fourier-prony recovery and prototype ADC,

    A. Bhandari, F. Krahmer, and T. Poskitt, “Unlimited sampling from theory to practice: Fourier-prony recovery and prototype ADC,” IEEE Transactions on Signal Processing , vol. 70, pp. 1131–1141, 2021

  14. [19]

    Wavelet- based reconstruction for unlimited sampling,

    S. Rudresh, A. Adiga, B. A. Shenoy, and C. S. Seelamantula, “Wavelet- based reconstruction for unlimited sampling,” in 2018 IEEE Inter- national Conference on Acoustics, Speech and Signal Processing (ICASSP). IEEE, 2018, pp. 4584–4588

  15. [20]

    Unlimited sampling of sparse signals,

    A. Bhandari, F. Krahmer, and R. Raskar, “Unlimited sampling of sparse signals,” in 2018 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP) . IEEE, 2018, pp. 4569–4573

  16. [21]

    Generalized approximate message passing for unlimited sampling of sparse signals,

    O. Musa, P. Jung, and N. Goertz, “Generalized approximate message passing for unlimited sampling of sparse signals,” in 2018 IEEE Global Conference on Signal and Information Processing (GlobalSIP) . IEEE, 2018, pp. 336–340

  17. [22]

    On the identifiability of sparse vectors from modulo compressed sensing measurements,

    D. Prasanna, C. Sriram, and C. R. Murthy, “On the identifiability of sparse vectors from modulo compressed sensing measurements,” IEEE Signal Processing Letters , vol. 28, pp. 131–134, 2020

  18. [23]

    On folded graph signals,

    F. Ji, P. Pratibha, and W. P. Tay, “On folded graph signals,” in2019 IEEE Global Conference on Signal and Information Processing (GlobalSIP) . IEEE, 2019, pp. 1–5

  19. [24]

    HDR imaging from quantization noise,

    A. Bhandari and F. Krahmer, “HDR imaging from quantization noise,” in 2020 IEEE International Conference on Image Processing (ICIP) , Oct 2020, p. 101–105

  20. [25]

    ITER-SIS: Robust unlimited sampling via iterative signal sieving,

    R. Guo and A. Bhandari, “ITER-SIS: Robust unlimited sampling via iterative signal sieving,” in ICASSP 2023 - 2023 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP) , Jun 2023, p. 1–5

  21. [26]

    Unlimited sampling theorem based on fractional fourier transform,

    H. Zhao and B.-Z. Li, “Unlimited sampling theorem based on fractional fourier transform,” Fractal and Fractional, vol. 7, no. 4, p. 338, 2023

  22. [27]

    Line spectral estimation via unlimited sampling,

    Q. Zhang, J. Zhu, F. Qu et al., “Line spectral estimation via unlimited sampling,” IEEE Transactions on Aerospace and Electronic Systems , 2024

  23. [28]

    Unlimited sampling beyond modulo,

    E. Azar, S. Mulleti, and Y . C. Eldar, “Unlimited sampling beyond modulo,” Applied and Computational Harmonic Analysis , vol. 74, p. 101715, 2025. [Online]. Available: https://www.sciencedirect.com/ science/article/pii/S1063520324000927

  24. [29]

    Residual recovery algorithm for modulo sampling,

    ——, “Residual recovery algorithm for modulo sampling,” in ICASSP 2022 - 2022 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP) , May 2022, p. 5722–5726

  25. [30]

    Lasso-based fast residual recov- ery for modulo sampling,

    S. B. Shah, S. Mulleti, and Y . C. Eldar, “Lasso-based fast residual recov- ery for modulo sampling,” in ICASSP 2023 - 2023 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP) , Jun 2023, p. 1–5

  26. [31]

    Modulo sampling of FRI signals,

    S. Mulleti and Y . C. Eldar, “Modulo sampling of FRI signals,” IEEE Access, vol. 12, pp. 60 369–60 384, 2024

  27. [32]

    A hardware prototype of wideband high-dynamic range analog- to-digital converter,

    S. Mulleti, E. Reznitskiy, S. Savariego, M. Namer, N. Glazer, and Y . C. Eldar, “A hardware prototype of wideband high-dynamic range analog- to-digital converter,” IET Circuits, Devices & Systems , vol. 17, no. 4, pp. 181–192, 2023

  28. [34]

    Sampling-50 years after shannon,

    M. Unser, “Sampling-50 years after shannon,” Proceedings of the IEEE, vol. 88, no. 4, pp. 569–587, 2000

  29. [35]

    Compressed sensing of analog signals in shift-invariant spaces,

    Y . C. Eldar, “Compressed sensing of analog signals in shift-invariant spaces,” IEEE Transactions on Signal Processing , vol. 57, no. 8, pp. 2986–2997, 2009

  30. [36]

    Unlimited dynamic range signal recovery for folded graph signals,

    F. Ji, W. P. Tay et al. , “Unlimited dynamic range signal recovery for folded graph signals,” Signal Processing, vol. 198, p. 108574, 2022

  31. [37]

    The structure of finitely gen- erated shift-invariant spaces in l2 (rd),

    C. Deboor, R. A. DeV ore, and A. Ron, “The structure of finitely gen- erated shift-invariant spaces in l2 (rd),” Journal of Functional Analysis , vol. 119, no. 1, pp. 37–78, 1994

  32. [38]

    Oblique dual frames and shift-invariant spaces,

    O. Christensen and Y . C. Eldar, “Oblique dual frames and shift-invariant spaces,” Applied and Computational Harmonic Analysis , vol. 17, no. 1, pp. 48–68, 2004

  33. [39]

    Nonuniform sampling and reconstruc- tion in shift-invariant spaces,

    A. Aldroubi and K. Gr ¨ochenig, “Nonuniform sampling and reconstruc- tion in shift-invariant spaces,” SIAM review, vol. 43, no. 4, pp. 585–620, 2001

  34. [40]

    Shift-invariant and sampling spaces asso- ciated with the fractional fourier transform domain,

    A. Bhandari and A. I. Zayed, “Shift-invariant and sampling spaces asso- ciated with the fractional fourier transform domain,” IEEE Transactions on Signal Processing , vol. 60, no. 4, pp. 1627–1637, 2011

  35. [41]

    Modulo sampling and recovery in shift- invariant spaces,

    Y . Kvich and Y . C. Eldar, “Modulo sampling and recovery in shift- invariant spaces,” inICASSP 2024 - 2024 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), 2024, pp. 11–15

  36. [42]

    Modulo sampling in shift-invariant spaces: Recovery and stability enhancement,

    ——, “Modulo sampling in shift-invariant spaces: Recovery and stability enhancement,” arXiv preprint arXiv:2406.10929 , 2024

  37. [43]

    From theory to practice: Sub-Nyquist sampling of sparse wideband analog signals,

    M. Mishali and Y . C. Eldar, “From theory to practice: Sub-Nyquist sampling of sparse wideband analog signals,” IEEE Journal of selected topics in signal processing , vol. 4, no. 2, pp. 375–391, 2010

  38. [44]

    Xampling: Analog to digital at sub-Nyquist rates,

    M. Mishali, Y . C. Eldar, O. Dounaevsky, and E. Shoshan, “Xampling: Analog to digital at sub-Nyquist rates,” IET circuits, devices & systems , vol. 5, no. 1, pp. 8–20, 2011

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Reviewed August 10, 2026 · model on record in the stance chip above.